| 00:00.49 | ManxPower | Druken, Um, move further south |
| 00:01.05 | ManxPower | In many parts of mexico you can live like a king of $30k/year |
| 00:02.12 | rajiv | how do you send a # to the called party of that is also used to transfer a call ? |
| 00:02.36 | kn0x | can anyone at all help me with my asterisk issues |
| 00:02.38 | kn0x | heres my sip.conf http://pastebin.ca/28725 no sip peers will register and im getting a "chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8155500 (len 398) to 69.90.155.70:-1 returned 5060: Bad file descriptor " on the cli |
| 00:03.02 | kn0x | running asterisk 1.2R2 |
| 00:03.16 | kn0x | on 2.6.13 |
| 00:03.19 | ManxPower | rajiv, I have no idea what you just said. |
| 00:03.42 | ManxPower | kn0x, did it fox it after you stopped and started Asterisk? |
| 00:03.52 | ManxPower | fox == fix |
| 00:04.33 | ManxPower | kn0x, what distro? |
| 00:04.47 | ManxPower | kn0x, REMOVE THE BINDADDR!!!!!!! |
| 00:05.07 | ManxPower | You can't bind to an address that isn't on the box |
| 00:05.32 | rajiv | lets say i have exten => _1NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN},,T) then pressing # will allow me to transfer the call. what if i want to send # to the other side and not have asterisk pay attention to it |
| 00:05.39 | litage | how does AMP (demo.coalescentsystems.ca) compare to ScopServ (scopserv.com)? |
| 00:06.05 | ManxPower | rajiv, you can't. |
| 00:06.35 | ManxPower | rajiv, # transfers are just a hack for devices that are too brain dead to provide proper transfer support. |
| 00:06.35 | kn0x | manx- gentoo |
| 00:07.10 | kn0x | im reloading after removal of bindaddr |
| 00:07.10 | ManxPower | kn0x, well you have your solution |
| 00:07.14 | rajiv | ManxPower: what do you mean by 'proper transfer support' ? like a transfer button on the phone ? |
| 00:07.15 | *** join/#asterisk slay (n=slay@ool-435143f3.dyn.optonline.net) |
| 00:07.15 | in-side | does anybody has libpthread working at FBSD 6.0 ? |
| 00:07.23 | ManxPower | rajiv, correct |
| 00:07.34 | rajiv | ManxPower: hmm. i have no transfer button on this phone. |
| 00:07.35 | ManxPower | in-side, try #asterisk-bsd |
| 00:07.39 | kn0x | manx- same issue after removal of bindaddr |
| 00:07.42 | ManxPower | rajiv, then get a real phone. 8-) |
| 00:07.48 | rajiv | there is 'flash' and 'function' |
| 00:07.55 | ManxPower | rajiv, What phone do you have? |
| 00:08.01 | rajiv | innomedia mta 3308 |
| 00:08.10 | ManxPower | rajiv, never heard of it. It's an IP phone? |
| 00:08.12 | rajiv | no one seems to use it. |
| 00:08.14 | rajiv | ya ip phone |
| 00:08.15 | Druken | asterboy: you still around? |
| 00:08.19 | rajiv | i had to add it to the wiki |
| 00:08.20 | in-side | ManxPower: does it exist?? |
| 00:08.33 | *** part/#asterisk slay (n=slay@ool-435143f3.dyn.optonline.net) |
| 00:08.37 | ManxPower | in-side, it did at one time. |
| 00:08.51 | rajiv | ManxPower: i think it is like an ATA with a built in speaker and handset. it's not a 'real' ip phone with multiple line buttons, or even a hold button |
| 00:09.00 | in-side | think every old geek at there had died thanks anyway ;) |
| 00:09.00 | ManxPower | in-side, there is also the asterisk-bsd mailing list |
| 00:09.08 | kn0x | manxpower- i changed the bindaddr to 0.0.0.0 |
| 00:09.12 | kn0x | and reloaded |
| 00:09.14 | kn0x | didnt fix it |
| 00:09.16 | ManxPower | rajiv, in the analog world "FLASH" is used for transfers |
| 00:09.19 | kn0x | same issues |
| 00:09.26 | ManxPower | kn0x, What part of "remove" did you not understand? |
| 00:09.33 | kn0x | okay |
| 00:09.48 | ManxPower | kn0x, did you ever have Asterisk working on this system? |
| 00:10.05 | ManxPower | kn0x, and do a unload chan_sip.so and then load chan_sip.so |
| 00:10.06 | kn0x | no |
| 00:10.16 | kn0x | i moved from slack to gentoo |
| 00:10.30 | Sedorox | good move |
| 00:10.31 | Sedorox | :p |
| 00:10.39 | ManxPower | kn0x, don't allow both alaw and ulaw |
| 00:10.43 | Sedorox | 'tho slack is still good for a lot of stuff |
| 00:11.00 | *** join/#asterisk Ropeguru (n=ropeguru@24.125.204.61) |
| 00:11.01 | kn0x | yeah |
| 00:11.06 | ManxPower | Sedorox, They are both good if you have more time than sense. |
| 00:11.08 | rajiv | ManxPower: k. i guess i'll remove T and use flash |
| 00:11.15 | Sedorox | hehe |
| 00:11.20 | ManxPower | rajiv, try it, if it works then great |
| 00:11.29 | kn0x | w00t manx- it registeered after i reloaded chan_sip.so |
| 00:11.31 | kn0x | ! |
| 00:11.37 | rajiv | flash seems to work okay for transfers |
| 00:11.44 | rajiv | but you are right. i need to get a better phone |
| 00:11.46 | Sedorox | redhat -> mandrake -> slack -> gentoo for me... over.. 6-7 years |
| 00:12.00 | ManxPower | We have used Mandrake for close to 8 years |
| 00:12.14 | ManxPower | In production since 6.2 |
| 00:12.19 | Sedorox | hehe |
| 00:12.27 | Sedorox | yea.. I started with Redhat 5.1 I think... |
| 00:12.29 | Sedorox | 5.1 or 5.2 |
| 00:12.30 | Sedorox | but eh |
| 00:12.37 | Sedorox | 'tis all in the learning experience |
| 00:14.00 | sahafeez | Sedorox: moved from gentoo to slack for the asterisk box. cleaner. just compile from source. fuckng ebuilds are screwed |
| 00:14.04 | sahafeez | for asterisk |
| 00:14.28 | rajiv | sahafeez: 1.0.x or 1.2.x ? |
| 00:14.44 | sahafeez | this was about 2 months ago |
| 00:14.47 | sahafeez | so 1.0 |
| 00:15.15 | rajiv | what kind of issues? i'm running 1.0.9 with no probs (and i'm a gentoo dev) |
| 00:15.32 | sahafeez | rajiv: i need stuff that was not in the ebuild. |
| 00:15.40 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 00:15.42 | klictel | hi all |
| 00:16.17 | rajiv | sahafeez: well we'll take patches to the ebuild if you have some. otherwise you know about the portage overlay support for rolling your own ebuilds? |
| 00:16.32 | sahafeez | rajiv: thats all. i wanted to build from HEAD and had issues under gentoo. got on the channel no one answered and i was in crunch so i just formated and went to slackware. i run gentoo for desktop on my sparc. |
| 00:16.49 | sahafeez | rajiv: yes. did not really have time to mess with it. |
| 00:17.11 | *** join/#asterisk fanguin (n=user@p548F5ED0.dip.t-dialin.net) |
| 00:17.16 | sahafeez | rajiv: i run freebsd mostly for servers anyway. |
| 00:17.19 | rajiv | k. the cvs ebuilds are/were busted |
| 00:19.57 | kn0x | Nov 14 12:19:05 WARNING[7458]: chan_zap.c:912 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
| 00:20.03 | kn0x | now im getting this with meetme |
| 00:20.30 | ManxPower | kn0x, do you have zaptel loaded and running? |
| 00:20.37 | kn0x | yes |
| 00:20.40 | kn0x | ofcourse |
| 00:20.43 | ManxPower | (either the card driver or ztdummy) |
| 00:20.45 | kn0x | ztdummy aswell |
| 00:20.50 | kn0x | no card |
| 00:20.56 | ManxPower | kn0x, and "lsmod" shows it's loaded |
| 00:21.09 | justinu | kn0x: what os? |
| 00:21.13 | kn0x | asterisk1*CLI> !modprobe -l /lib/modules/2.6.13-gentoo-r5/misc/ztdynamic.ko /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko /lib/modules/2.6.13-gentoo-r5/misc/ztd-loc.ko /lib/modules/2.6.13-gentoo-r5/misc/ztd-eth.ko /lib/modules/2.6.13-gentoo-r5/misc/zaptel.ko |
| 00:21.16 | kn0x | gentoo |
| 00:21.24 | justinu | does that use udevd? |
| 00:21.30 | kn0x | runnign 2.6.13r5(gentoo) kernel |
| 00:21.37 | kn0x | udevd |
| 00:21.39 | skyen | yes, gentoo uses udevd |
| 00:21.50 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 00:21.52 | justinu | did you modify the files like README.udev told you to? |
| 00:22.07 | ManxPower | kn0x, and "lsmod" shows it's loaded? |
| 00:22.17 | kn0x | never came across that justinu |
| 00:22.28 | ManxPower | kn0x, Um, the modules must be loaded BEFORE Asterisk is started. |
| 00:22.41 | justinu | kn0x: look in the zaptel directory and follow those instructions to create the /dev/zap files |
| 00:22.56 | kn0x | heres the thing though i cannot load zaptel |
| 00:22.58 | kn0x | FATAL: Error inserting zaptel (/lib/modules/2.6.13-gentoo-r5/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 00:23.06 | kn0x | modproble zaptel |
| 00:23.21 | justinu | modprobe ztdummy, since you don't have a zaptel board |
| 00:23.23 | ManxPower | kn0x, if zaptel and ztdummy are not BOTH loaded, you can't use meetme |
| 00:23.42 | kn0x | i did the CONFIG_CRC_CCITT=y |
| 00:23.47 | kn0x | in the menuconfig |
| 00:23.52 | ManxPower | justinu, modprobe ztdummy will autoload zaptel |
| 00:23.59 | justinu | ok |
| 00:24.12 | kn0x | dmesg shows |
| 00:24.16 | kn0x | (flood) |
| 00:24.17 | kn0x | zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol zt_receive ztdummy: Unknown symbol zt_transmit ztdummy: Unknown symbol zt_unregister ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol zt_register ztdummy: Unknown symbol rtc_control |
| 00:24.37 | ManxPower | kn0x, sounds like you need to modporbe rtc and crc_citt |
| 00:25.08 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 00:25.08 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 00:25.10 | kn0x | those are both built into the kernel |
| 00:25.25 | kn0x | i told them both to build inside the kernel |
| 00:25.47 | rajiv | did you install and reboot with the new kernel ? |
| 00:25.57 | kn0x | ofcourse |
| 00:26.54 | ManxPower | kn0x, ztdummy MAY reuiqre they are built as modules |
| 00:27.06 | kn0x | ! |
| 00:27.46 | LostFrog | Hmm.. the transfer on hook setting on snoms can be dangerous. |
| 00:28.09 | ManxPower | <PROTECTED> |
| 00:28.09 | ManxPower | On Time |
| 00:28.12 | kn0x | do i need extended rtc |
| 00:28.13 | kn0x | ? |
| 00:28.22 | ManxPower | Yup, for UPS a package being ontime IS an exception |
| 00:28.45 | ManxPower | kn0x, I dunno. I've never been so cheap that I can't buy a $9 X100P clone if I need to use MettMe |
| 00:29.09 | LostFrog | lol |
| 00:30.32 | ManxPower | I got a request from a client to change the message when someone tries to hit "0" in the IVR from "No, no, no! Try again smart guy!". They wanted something "more professional" |
| 00:30.46 | sahafeez | ~rfc2833 |
| 00:31.21 | LostFrog | lol.. MettMe |
| 00:31.38 | LostFrog | Danm snom for their lack of documentation and their subtleties. |
| 00:31.58 | LostFrog | :) |
| 00:32.03 | sahafeez | ~progressinband |
| 00:32.38 | LostFrog | I can't understand why, ManxPower.. |
| 00:32.54 | Anthro | So I'm doing make INSTALL_PREFIX=/usr/local ...is that the right way to go about it? |
| 00:33.00 | sahafeez | which is better dtmfmode=inband |
| 00:33.01 | sahafeez | or rfc2833 |
| 00:33.05 | LostFrog | Maybe 'you-sound-cute', ManxPower? |
| 00:33.10 | LostFrog | rfc2833 |
| 00:33.23 | ManxPower | LostFrog, it's an Allison recording. |
| 00:33.26 | LostFrog | Ever try inband over g729 or g726? :) |
| 00:33.36 | LostFrog | ManxPower: I know. i was joking. |
| 00:33.38 | ManxPower | I didn't think THAT many people tried to 0 out of the IVR |
| 00:33.49 | LostFrog | So is 'you-sound-cute' |
| 00:33.55 | LostFrog | or 'you-sound-impatient' |
| 00:34.13 | LostFrog | you-seem-impatient, even. |
| 00:34.14 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 00:35.31 | ManxPower | LostFrog, Apparently their clients don't have a sense of humor. The may me take off the message at the end of the IVR options. "If you don't know your party's extension or are simply confused, stay on the line and someone will be with you shortly" |
| 00:35.51 | ManxPower | Apparently some little old lady that called was offended by it |
| 00:36.11 | Sedorox | sahafeez: I don't see any problems with the ebuilds right now.. 'tho when I see one.. I'm gonna do source |
| 00:36.18 | mog_work | lol |
| 00:36.19 | LostFrog | How?? That is nonoffensive. |
| 00:36.21 | mog_work | thats sad manx |
| 00:36.25 | sahafeez | Sedorox: this was a month ago |
| 00:36.36 | ManxPower | mog_work, My client loved it, their clients didn't. |
| 00:36.42 | Sedorox | ah |
| 00:37.05 | Sedorox | I'm running 1.0.9 from ebuild right now on a production box.. doesn't seem to have any problems *knocks on wood* |
| 00:37.12 | mog_work | i often have you are not the next caller in line on my phone |
| 00:37.26 | ManxPower | I don't work for people without a twisted sense of humor and medical people DO have a twisted sense of humor. |
| 00:37.36 | LostFrog | Now.. if it said 'Well.. obviously, you are clueless, let me connect you to the operator.', I would understand. |
| 00:39.09 | kn0x | manx- hold with me i have to eat |
| 00:39.41 | LostFrog | What is an 'intonation file?' |
| 00:40.13 | ManxPower | It's looking like the thread '"open" asterisk' is going to be the Thread From Hell |
| 00:40.58 | LostFrog | I guess I should subscribe to the mailing lists. |
| 00:41.10 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:41.21 | mog_work | yeah manx |
| 00:41.26 | litage | does AMP support Asterisk 1.2 yet? |
| 00:41.33 | mog_work | royk tends to do things like that |
| 00:42.20 | MikeJ[Laptop] | royk tends o do things like what? |
| 00:42.25 | *** join/#asterisk xiel (n=xiel@pcp08477368pcs.summit01.tn.comcast.net) |
| 00:42.32 | mog_work | troll |
| 00:42.39 | mog_work | spread FUD |
| 00:42.45 | mog_work | whatever you want to call it |
| 00:42.47 | MikeJ[Laptop] | YAY FUD! |
| 00:42.50 | LostFrog | mmm.. FUD |
| 00:42.52 | ManxPower | MikeJ[Laptop], Some people involved with Asterisk are ..... high strung. |
| 00:42.54 | MikeJ[Laptop] | what list? |
| 00:43.00 | MikeJ[Laptop] | hehe |
| 00:43.00 | mog_work | users |
| 00:43.01 | MikeJ[Laptop] | yes... |
| 00:43.10 | MikeJ[Laptop] | oh... I havn't read users in sooo long |
| 00:43.11 | MikeJ[Laptop] | no time.. |
| 00:43.22 | MikeJ[Laptop] | have a hard enough time keeping up w/ dev and cvs |
| 00:43.31 | mog_work | some people dont like it when they say their employeer engages in anti-competive biz tactics |
| 00:43.43 | mog_work | esp when they are baseless |
| 00:44.03 | MikeJ[Laptop] | and I have been working on some other stuff in msvc lately... I haven't touched a gcc based project in tooo long |
| 00:44.25 | LostFrog | ok.. off-topic, anyone have any recommendations for linux webmail servers? |
| 00:44.37 | ManxPower | mog_work, here is a case where Mark could publicly post to the mailing list stating Digium's official position on the issue. |
| 00:44.41 | MikeJ[Laptop] | LostFrog, sure somone does |
| 00:44.43 | MikeJ[Laptop] | <PROTECTED> |
| 00:44.53 | LostFrog | Yeah, funny, MikeJ. |
| 00:44.54 | LostFrog | :) |
| 00:44.54 | mog_work | yeah i am gonna buzz him next time i see him |
| 00:45.11 | mog_work | im sure there will be words |
| 00:45.16 | MikeJ[Laptop] | ok.. back to my msvc dependency lib download\compile bat file work :P |
| 00:45.39 | MikeJ[Laptop] | mog_work, that'll teach you to read the users list |
| 00:45.42 | *** part/#asterisk SplasPood (n=sp@paravolve.net) |
| 00:45.42 | LostFrog | ewww.. msvc. |
| 00:45.49 | *** join/#asterisk bweschke (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 00:46.17 | MikeJ[Laptop] | LostFrog, honestly, I like the debug system... I am not in love with the project\make system |
| 00:46.34 | MikeJ[Laptop] | but that being said... not like makefiles or autotools are any good either |
| 00:47.16 | LostFrog | I don't want to browse through freshmeat again. :( |
| 00:47.24 | MikeJ[Laptop] | if you have good portable code, msvc is fast, makes nice and little files, and rebuilds\relinks 10 times as fast as gcc ever has |
| 00:48.01 | MikeJ[Laptop] | now the trickery part there is the portable code part. |
| 00:48.25 | MikeJ[Laptop] | as MSVC does not have any posix stuff... just ansi c++ |
| 00:48.30 | LostFrog | hmm.. I hate companies that don't post prices for their software.. |
| 00:49.06 | MikeJ[Laptop] | if microsoft built real posix compliant api's into their system stock, it would be way way easier |
| 00:49.10 | LostFrog | I would pay a decent amount for a good webmail software. |
| 00:49.23 | MikeJ[Laptop] | LostFrog, MS exchange :D |
| 00:49.29 | tzanger | LostFrog: openwebmail or squirrelmail (what I use) isn't good? |
| 00:49.30 | MikeJ[Laptop] | hehe |
| 00:49.32 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-57.cybersurf.com) |
| 00:49.43 | bweschke | LostFrog: I like and use squirrelmail |
| 00:49.49 | bweschke | it's not perfect, but can't beat the price |
| 00:49.49 | bweschke | :) |
| 00:49.54 | mog_work | squirelmail is awesome |
| 00:50.09 | kn0x | manxpower- asterisk1 ~ # modprobe rtc FATAL: Module rtc not found. |
| 00:50.12 | tzanger | I'd use openwebmail if it didn't try and access the mail directly (i.e. use the damn imap server) |
| 00:50.46 | rayvd | Barns. |
| 00:51.13 | LostFrog | I will have to try both. |
| 00:51.25 | LostFrog | After I see a man about a horse. |
| 00:52.20 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 00:53.34 | stbain | number 3 to place in the fourth? |
| 00:54.46 | Flauto | Nov 14 18:31:17 WARNING[22778]: chan_sip.c:9575 handle_response_register: Got 200 OK on REGISTER that isn't a register, what does this mean |
| 00:55.21 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 00:55.45 | LostFrog | Number 10 in the 2nd. |
| 00:55.49 | marcus2 | so, is anyone using chan_bluetooth? =D |
| 00:55.51 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 00:57.32 | LostFrog | lol.. webmail for nuts! |
| 00:58.40 | nick125 | marcus2: that exists? :o |
| 00:58.46 | marcus2 | it appears to |
| 00:59.00 | nick125 | weird |
| 00:59.01 | marcus2 | the idea is certainly very cool |
| 00:59.07 | marcus2 | http://www.crazygreek.co.uk/content/chan_bluetooth |
| 00:59.10 | nick125 | "The first version of chan_bluetooth has been released. This code allows you to use a bluetooth compatible cell phone to connect to your Asterisk box." |
| 01:01.03 | mog_work | thats hott |
| 01:01.39 | d-tech | cell phone with a fifty foot operational radius ... novel concept?! |
| 01:01.49 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 01:01.59 | YoMama | anyone here have issues with the S100U? |
| 01:02.27 | ManxPower | YoMama, You mean the IAXy? |
| 01:02.48 | mog_work | no s100u predates iaxy |
| 01:02.53 | mog_work | usb ata |
| 01:03.01 | YoMama | ManxPower: no...the S100U...usb fxs thingamajigger |
| 01:03.11 | YoMama | the fact that no one usually knows what i'm talking about means it's a serious piece of crap |
| 01:03.18 | RoyK | mog_work: the s100u is no good |
| 01:03.29 | ManxPower | YoMama, They were so unreliable, flakey, and died from static that digium stopped selling them like a year ago |
| 01:03.42 | mog_work | yeah digium had bad run with them |
| 01:03.44 | YoMama | ManxPower: uh huh...figured that out.. :( |
| 01:03.45 | *** part/#asterisk Ropeguru (n=ropeguru@24.125.204.61) |
| 01:03.47 | mog_work | that and x100p |
| 01:03.56 | YoMama | i have an x100p...seems to work great |
| 01:03.58 | mog_work | otherwise we make and design hw ourselves |
| 01:04.21 | ManxPower | YoMama, Digium didn't design the X100P or the S100U (afaik) |
| 01:04.32 | YoMama | i should just stop being such a cheapass and buy a TDM400 |
| 01:04.42 | mog_work | maybe |
| 01:04.47 | ManxPower | YoMama, or be less of a cheapass and get a SIPura ATA |
| 01:04.54 | mog_work | yup manx we wrote drivers for themn |
| 01:04.59 | mog_work | but they are normal devices |
| 01:05.11 | YoMama | ManxPower: i also have an old Motorola VT1005V from Vonage...i think i gotta pay them to unlock it though |
| 01:05.38 | YoMama | ManxPower: i hear some of those are locked..gotta be careful what u buy on ebay |
| 01:05.39 | d-tech | [Digium didn't design the X100P] ... huh? |
| 01:05.49 | ManxPower | YoMama, Katrina took care of all that sort of stuff that I could not bear to throw away, but wasn't very usable. It was all sitting on the floor. |
| 01:06.03 | mog_work | nope x100p is a stock winmodem |
| 01:06.15 | mog_work | we used to say it on site somewhere |
| 01:06.21 | mog_work | everything else is designed here though |
| 01:06.21 | YoMama | ManxPower: u are from lousiana? |
| 01:06.23 | ManxPower | d-tech, The X100P is pretty much a specific type of winmodem, using a specific chipset and specific firmware that Digium wrote zaptel drivers for. |
| 01:06.32 | ManxPower | YoMama, Lived near Gulfport MS |
| 01:06.45 | ManxPower | 1/2 mile from the ocean |
| 01:06.51 | YoMama | ManxPower: wow |
| 01:06.55 | d-tech | they just re-marketed it?! |
| 01:07.06 | IronHelix | i would be surprised if Vonage allowed any sort of vt1xxx unlcokign |
| 01:07.06 | mog_work | yes |
| 01:07.11 | ManxPower | d-tech, they added a heat sink and I think took off a resisotr |
| 01:07.23 | YoMama | IronHelix: why? i've read reports they'll unlock it for $15 |
| 01:07.33 | IronHelix | a vt1000? surprising |
| 01:07.45 | IronHelix | mostly because there are no config fields |
| 01:07.45 | ManxPower | YoMama, 90% of the city I lived in was TOTALLY destroyed. The building I was renting only had minor wind damage and only 3ft of water in it. |
| 01:07.48 | YoMama | IronHelix: in fact..i'm gonna call 'em now |
| 01:08.01 | YoMama | ManxPower: the company i used to work for had an office in NOLA |
| 01:08.02 | ManxPower | I lived in "Waveland, MS" |
| 01:08.06 | sahafeez | issue: anyone hits # it does -- Playing 'pbx-transfer' (language 'en'). can i undo that |
| 01:08.11 | YoMama | it was on the 20th floor, but the windows were all blownout |
| 01:08.21 | ManxPower | sahafeez, take off T and t from the Dial command. |
| 01:08.36 | ManxPower | YoMama, my largest client's HQ is north of NOLA, but had some small offices in the city. |
| 01:08.49 | IronHelix | i would be surprised if they do... the vt1000 unlike others like the pap2 is solely designed as provider CPE, to be delivered locked. To the best of my knowledge there is no 'useful' web config screen and it wont let you change the tftp server |
| 01:08.54 | IronHelix | of course i could be wrong |
| 01:08.55 | sahafeez | ManxPower: and that will not affect the ability to transfer a call with the "transfer button" |
| 01:09.03 | ManxPower | sahafeez, NO!~ |
| 01:09.04 | *** join/#asterisk ptiggerdine (n=ptiggerd@c210-49-98-194.rochd1.qld.optusnet.com.au) |
| 01:09.07 | IronHelix | yomama LMK what happensw with it |
| 01:09.15 | ManxPower | t and T are ONLY for DTMF (#) transfers |
| 01:09.15 | sahafeez | ManxPower: you rule |
| 01:09.32 | sahafeez | ok, so no T or t |
| 01:09.40 | sahafeez | is RFC2833 DTMF |
| 01:09.53 | sahafeez | yes..answer my own question sorry |
| 01:10.07 | RoyK | hehehehehe |
| 01:10.11 | IronHelix | RFC/sipinfo/inband all ways of sending dtmf over SIP |
| 01:10.19 | RoyK | digium is scared |
| 01:10.24 | IronHelix | ? |
| 01:10.31 | RoyK | they have started sensoring the list |
| 01:10.40 | IronHelix | seriously? |
| 01:10.42 | IronHelix | url? |
| 01:10.43 | RoyK | yes |
| 01:10.52 | IronHelix | / more info? |
| 01:11.02 | pcm | yeah ! I got banned :) |
| 01:11.18 | IronHelix | for what? |
| 01:11.21 | RoyK | i just replied to a post of my initial post of [Asterisk-Users] "open" asterisk? |
| 01:11.40 | RoyK | and got an answer "post to a moderated list" |
| 01:11.46 | RoyK | never got that before |
| 01:12.01 | tzanger | huh? |
| 01:12.41 | RoyK | they prolly don't want to hear people saying things against their own 'reason' |
| 01:13.00 | IronHelix | i didnt even know there was such a thing as a tdm400 clone |
| 01:13.11 | pcm | well you don't read the list :) |
| 01:13.23 | IronHelix | dont have enough time :( |
| 01:13.35 | tzanger | I could never get to the clone mfg's website |
| 01:13.39 | pcm | I don't read the list either ... only the titles from the digest |
| 01:13.41 | tzanger | I guess Digium's controling my DNS |
| 01:13.50 | ManxPower | RoyK, Im, I get held for moderation occsionally. The mailing list software does it automatically when it thinks it should. |
| 01:14.01 | ManxPower | IronHelix, there isn't |
| 01:14.08 | YoMama | IronHelix: i was gonna do vonage..except those ijits could never get my phone # transferred |
| 01:14.11 | RoyK | ManxPower: we'll see |
| 01:14.11 | YoMama | so i said screw it |
| 01:14.23 | pcm | manxpower: how to unban myself from the list ? |
| 01:14.34 | ManxPower | pcm, unban who from what list? |
| 01:14.46 | pcm | manxpower: well I can't post to the list anymore :) |
| 01:14.49 | RoyK | ManxPower: it was very timey, just when i posted something that was in the way of asterisk |
| 01:15.20 | ManxPower | RoyK, I vaguely recall it happened to me when I had a tiny post or a reply with no new content |
| 01:15.32 | IronHelix | yomama- same. tried to have them transfer, it would take 20days, 2 years later nothing |
| 01:15.36 | ManxPower | pcm, Perhaps you did something to piss someone off. |
| 01:15.43 | IronHelix | now im trying to get them to get their claws off my 'temporary number' |
| 01:15.45 | ManxPower | It IS Digium's list, afterall. |
| 01:16.02 | IronHelix | hmmm, just read the thread, this looks like a mess :( |
| 01:16.02 | puzzled | evening all |
| 01:16.10 | RoyK | ManxPower: ok |
| 01:16.12 | *** join/#asterisk klydal (n=willow_8@ip68-107-201-231.nc.hr.cox.net) |
| 01:16.16 | ManxPower | IronHelix, I'm adding it to my killfile |
| 01:16.23 | RoyK | well |
| 01:16.25 | RoyK | after all |
| 01:16.31 | RoyK | asterisk is pure open source |
| 01:16.36 | *** join/#asterisk marc324 (n=marc3234@206-248-152-219.dsl.teksavvy.com) |
| 01:16.37 | RoyK | freedom to |
| 01:16.38 | RoyK | eh |
| 01:16.40 | RoyK | digium? |
| 01:16.44 | IronHelix | so if i got it right, openpbx guys hired allison to record shit, mark said no way, so allison has to choose between openpbx and digium |
| 01:16.47 | marc324 | ne1 familiar with ser config? |
| 01:16.48 | IronHelix | or did i miss something |
| 01:16.57 | ManxPower | RoyK, you either need to get laid or get some valium |
| 01:17.11 | YoMama | oh my god...their customer care people are idiots |
| 01:17.17 | ManxPower | IronHelix, that is the claim by some people. I'm just waiting. |
| 01:17.17 | YoMama | thank god i cancelled |
| 01:17.20 | klydal | hey everyone, Im new. Im confused about what asterisk is. Is it a Voip? or just extra service? |
| 01:17.41 | RoyK | ManxPower: my short little test message also got blocked |
| 01:17.46 | tzanger | klydal: asterisk is an open-source PBX |
| 01:17.47 | RoyK | strange thing.... |
| 01:17.53 | IronHelix | klydal- asterisk is a piece of software that acts like a pbx or phone system. it can work with voip or any other voice channel pretty much |
| 01:18.00 | ManxPower | klydal, it's a PBX that supports several ways of getting/sending phone calls. |
| 01:18.01 | tzanger | it can do VOIP, it can do analog lines, it can do T1/E1 and PRI |
| 01:18.06 | ManxPower | RoyK, did it say "test message"? |
| 01:18.07 | tzanger | it's a phone system |
| 01:18.14 | puzzled | klydal: go to asteriskdocs.org and read all about it in the book online or buy the book |
| 01:18.19 | klydal | so its free? |
| 01:18.19 | ManxPower | ~docs |
| 01:18.21 | jbot | methinks docs is 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 |
| 01:18.21 | tzanger | puzzled++ |
| 01:18.25 | Math` | klydal: yeah |
| 01:18.32 | ManxPower | klydal, the software is, obviously the hardware is not. |
| 01:19.00 | YoMama | hmm..what's the diff between asterisk and openpbx? |
| 01:19.00 | puzzled | ManxPower: and that should obviously change :) |
| 01:19.11 | puzzled | YoMama: the name? |
| 01:19.12 | *** join/#asterisk test34- (n=test34@unaffiliated/test34) |
| 01:19.12 | ManxPower | YoMama, politics |
| 01:19.21 | tzanger | YoMama: not too much right now. Mostly political |
| 01:19.23 | mog_work | it will diverge a lot, just hasnt yet |
| 01:19.26 | RoyK | ManxPower: no |
| 01:19.34 | YoMama | mog_work: what's their philosophy? |
| 01:19.36 | YoMama | hey math |
| 01:19.40 | Math` | hey |
| 01:19.41 | klydal | thanks guys |
| 01:19.51 | mog_work | well it depends who you ask yomama |
| 01:19.57 | RoyK | also |
| 01:19.58 | mog_work | some people say its the truely open pbx |
| 01:20.02 | ManxPower | YoMama, go to openpbx.org to learn about it. |
| 01:20.03 | YoMama | why diverge...it's not like someone can't write zaptel compatible drivers for their hardware |
| 01:20.07 | mog_work | as they dont do dual licenseing |
| 01:20.13 | mog_work | and arent part of digium |
| 01:20.21 | RoyK | a message with subject "compilation faliure" got blocked |
| 01:20.22 | mog_work | others say they are really really angry at digium |
| 01:20.29 | puzzled | klydal: the book gives you a lot of info as does voip-info.org and the mailinglist archives. best is to get your hands dirrty, install play, learn and conquer |
| 01:20.36 | mog_work | it seems like a bit of both |
| 01:20.38 | ManxPower | RoyK, you really need to stop channeling bkw_ |
| 01:20.39 | RoyK | ManxPower: someone @digium is prolly pissed off at me |
| 01:20.52 | RoyK | ManxPower: i've done that once |
| 01:20.54 | ManxPower | RoyK, Honestly, why should we care? |
| 01:20.58 | klydal | so I could use asterik instead of geting something like vonage? |
| 01:21.00 | RoyK | we? |
| 01:21.06 | tzanger | RoyK: I dunno I've said some pretty sharp criticisims of digium on -users and not gotten banned |
| 01:21.14 | ManxPower | RoyK, Well the people on this channel, for example. |
| 01:21.17 | tzanger | RoyK: that tends to tell me that they don't moderate those who don't toe the line |
| 01:21.21 | YoMama | mog_work: hmm...that's really too bad...time would be better spent improving asterisk |
| 01:21.25 | YoMama | digium only sells software |
| 01:21.27 | *** join/#asterisk cmslaght (n=cmslaght@admin.ambt.net) |
| 01:21.28 | YoMama | err..hardware |
| 01:21.29 | ManxPower | YoMama, I agree. |
| 01:21.30 | mog_work | the only thing that has been removed roy |
| 01:21.32 | tzanger | YoMama: no |
| 01:21.34 | tzanger | they sell ABE too |
| 01:21.35 | mog_work | is gpl violation g729 links |
| 01:21.42 | YoMama | ABE? |
| 01:21.42 | RoyK | it's a new thing :) |
| 01:21.46 | cmslaght | Has anyone messed with vmexten in the sip.conf? |
| 01:21.55 | RoyK | suddenly digium starts censoring |
| 01:21.58 | mog_work | digium sells hw , sw, and services |
| 01:22.03 | ManxPower | mog_work, I noticed that when I was trying to find the URL for the pirate G729 stuff so I could report the site to the patent holders |
| 01:22.07 | puzzled | cmslaght: new to me |
| 01:22.07 | CoaxD | God damnit. DiscoverCard's login thing is being shittttttt slow |
| 01:22.21 | mog_work | yeah we have to do it manx or we lose are license |
| 01:22.28 | YoMama | mog_work: k...but my point is nothing is stopping anyone from writing zaptel compatible drivers right? |
| 01:22.30 | *** join/#asterisk bweschke (n=bweschke@204.96.162.40) |
| 01:22.34 | mog_work | nothing |
| 01:22.35 | ManxPower | mog_work, It was annoying, but understandable |
| 01:22.43 | mog_work | and several none digium cards are supported yomama |
| 01:22.45 | tzanger | mog_work: when digium *does* censor, is themessage replaced with "this message was removed due to patent/copyright violation material" ? |
| 01:22.50 | puzzled | ManxPower: only "pirate" in countries where the patents are valid... |
| 01:22.52 | mog_work | yes tzanger |
| 01:23.01 | tzanger | mog_work: that's excellent |
| 01:23.03 | mog_work | or something like that |
| 01:23.24 | YoMama | mog_work: k..then how is it not open source? if i was "mad" at digium...i'd go off and start my own hardware company that made these boards and write drivers for them |
| 01:23.40 | ManxPower | YoMama, If you want your code included with Asterisk, Digium requires you to give them a free license to use your code and reserve the right to sell a non-opensource product (Asterisk Business Edition) and use your code. Some people don't like that and so OpenPBX was born. |
| 01:24.03 | mog_work | some people thought abe was digium's way to get rich of devs hard work |
| 01:24.11 | YoMama | ManxPower: ah |
| 01:24.17 | mog_work | they also didnt like disclaimer |
| 01:24.22 | mog_work | or the structure of dev model |
| 01:24.35 | YoMama | ManxPower: screw that then...just supply it all as patches then :) |
| 01:24.50 | mog_work | as asterisk is lead by digium employee mark |
| 01:24.54 | YoMama | ManxPower: not the most graceful...but it works |
| 01:25.03 | ManxPower | YoMama, you will hear people that say Digium requires you to sign your copyright on your code over to Digium. Yes, you CAN do that, but you can also just give them a free unrestricted license to use your code and you keep the copyright. |
| 01:25.23 | drumkilla | Digium does not even provide a disclaimer to do that. |
| 01:25.30 | mog_work | not yet |
| 01:25.31 | tzanger | ? |
| 01:25.35 | mog_work | but it has been discussed |
| 01:25.37 | drumkilla | indeed |
| 01:25.39 | tzanger | Digium's disclaimer doesn't assign copyright |
| 01:25.47 | puzzled | aren't there 2 disclaimers, a short one and long one? |
| 01:25.50 | drumkilla | the FSF, for example, requires you to assign copyright to them |
| 01:25.51 | Math` | yup |
| 01:25.57 | tzanger | just perpetual transferable license to use your code |
| 01:25.59 | tzanger | you keep the copyright |
| 01:26.01 | mog_work | yeah |
| 01:26.10 | mog_work | both options are essentially that now |
| 01:26.16 | mog_work | one is more legalize than the other |
| 01:26.19 | ManxPower | drumkilla, I think they refer to: http://www.digium.com/disclaim.changes |
| 01:26.19 | drumkilla | one puts your code in public domain, the other just provides Digium with an unlimited license, but you retain copyright ... |
| 01:26.34 | YoMama | ManxPower: huh...so if i write something nifty for asterisk...and want it included...basically i'm allowing them to use it for free in their commercial edition |
| 01:26.41 | mog_work | yes |
| 01:26.42 | ManxPower | Where I prefer this one: http://www.digium.com/disclaimer.txt Hell, I even amended this disclaimer to make some changes. |
| 01:26.44 | mog_work | but you dont have to do that |
| 01:26.50 | mog_work | you can keep your patch up by yourself |
| 01:26.52 | ManxPower | YoMama, Yes. |
| 01:26.55 | mog_work | and people can apply patch |
| 01:27.12 | mog_work | like for example my patch res_xmpp isnt in tree yet |
| 01:27.17 | mog_work | so i maintain it elsewhere |
| 01:27.20 | drumkilla | but if you get your patch in the tree, we update it for you. |
| 01:27.22 | Math` | and you have to re-diff it every week :P |
| 01:27.23 | mog_work | till its ready |
| 01:27.31 | mog_work | more often than that math |
| 01:27.37 | Math` | yeah |
| 01:27.38 | mog_work | as it is few thousand lines now |
| 01:27.40 | puzzled | so what do you guys think about merging e.g. the openpbx.org autoconf stuff into * (which obviously means the abe can only use the parts that are disclaimed)? |
| 01:27.57 | tzanger | puzzled: wont' happen for many reasons already |
| 01:28.04 | RoyK | seems digium has suddenly moderated the -users list |
| 01:28.06 | tzanger | basically nobody at digium's willing to maintain it |
| 01:28.12 | RoyK | all my messages are being held |
| 01:28.12 | ManxPower | puzzled, Are you volunteering to maintain it. |
| 01:28.14 | tzanger | RoyK: shut up about that already |
| 01:28.19 | drumkilla | RoyK: you have no idea what you are talking about |
| 01:28.22 | tzanger | you've probably been shitlisted for one reason or another |
| 01:28.23 | mog_work | that and no code goes into main tree unless its disclaimed |
| 01:28.27 | drumkilla | the list server gets overloaded occassionally |
| 01:28.28 | sivana | heh |
| 01:28.29 | RoyK | drumkilla: ??? |
| 01:28.32 | RoyK | tzanger: ??????? |
| 01:28.32 | drumkilla | takes some time for messages to get through |
| 01:28.33 | mog_work | i will go look into it roy |
| 01:28.36 | puzzled | ManxPower: that's an open door answer. I was looking for an open discussion |
| 01:28.38 | drumkilla | RoyK: you are making stuff up |
| 01:28.39 | mog_work | but the only person who could do that |
| 01:28.43 | mog_work | isnt around at the moment |
| 01:28.49 | tzanger | RoyK: have you been making tons of posts in the last [short amount of time] ? |
| 01:28.49 | mog_work | and would not be around to do so |
| 01:28.58 | RoyK | drumkilla: no. i'm not. wanna see the emails i just got? |
| 01:29.10 | RoyK | i've been posting loads of stuff to that list |
| 01:29.13 | RoyK | but just now |
| 01:29.18 | YoMama | vonage sucks...they can't even hear me.. |
| 01:29.19 | RoyK | all gets 'moderated' |
| 01:29.26 | ManxPower | puzzled, I know, but it's kind of silly to talk about it if someone isn't going to maintain it. |
| 01:29.26 | RoyK | that has never happened before |
| 01:29.28 | RoyK | NEVER |
| 01:29.30 | drumkilla | do you think anyone at digium has time to moderate that list? |
| 01:29.45 | tzanger | RoyK: what I am getting at is this: Have you posted a ton of messages in the last little while ... could you have been misdetected as a runaway MUA? |
| 01:29.46 | drumkilla | well, the answer is absolutely not ... |
| 01:29.49 | RoyK | drumkilla: perhaps they just put my address on a bad-guy-list |
| 01:29.55 | ManxPower | drumkilla, only to unsubscribe spammers or people that stir up trouble 8-) |
| 01:30.03 | RoyK | tzanger: not in a little while |
| 01:30.12 | RoyK | tzanger: not more than earlier |
| 01:30.24 | puzzled | ManxPower: agree but was looking for ideas/opinions on the viability of such an idea before the pratical part needed to be addressed |
| 01:30.27 | mog_work | it is moderated from time to time, for g729 violations |
| 01:30.42 | ManxPower | The whole "open" Asterisk thread has about as much useful stuff as the GPL threads or the G723.1/G729 threads and the patent threads. |
| 01:30.45 | mog_work | but the person who can moderate is on a trip, and would not be online to do so |
| 01:31.04 | RoyK | ManxPower: wrong |
| 01:31.15 | puzzled | mog_work: that must be a challenge given the fact that the site that has the stuff is (afaik) not subject to those US patents |
| 01:31.18 | RoyK | ManxPower: the thread is about digium abusing their position |
| 01:31.26 | ManxPower | puzzled, I don't think Digium has any interest in getting patches from OpenPBX.org. Since one of OpenPBX.org's issues is disclaiming source. |
| 01:31.28 | mog_work | we only remove links to binary |
| 01:31.34 | mog_work | that is our legal need |
| 01:31.49 | ManxPower | RoyK, Um, it's their mailing list, it's their product, they CAN'T abuse their position. |
| 01:31.51 | tzanger | http://i.a.cnn.net/cnn/2005/US/11/14/parents.slain/story.borden.police.jpg |
| 01:31.54 | tzanger | I can't believe that chick's 14 |
| 01:32.03 | ManxPower | That's like saying I'm abuseing my position if I don't let you use my credit card. |
| 01:32.04 | mog_work | yikes |
| 01:32.11 | Math` | tzanger: </outofcontext> ? lol :P |
| 01:32.16 | tzanger | I'm *so* glad I've got a stable relationship... I'd be in jail for sure |
| 01:32.23 | Math` | lol |
| 01:32.25 | RoyK | ManxPower: i'm talking about the thread. that is about digium abusing their 'monopoly' |
| 01:32.26 | tzanger | or maybe not |
| 01:32.27 | Anthro | LostFrog: How did you compile 1.2rc2 on Debian stable? It seems to need a newer libpri. |
| 01:32.28 | tzanger | http://i.a.cnn.net/cnn/2005/US/11/14/parents.slain/story.suspect.handout.jpg |
| 01:32.30 | puzzled | ManxPower: yup but perhaps its (in the future) ppl can discuss and ammend their differences so both communities can benefit from the work from all sides |
| 01:32.31 | tzanger | that's the dude's she's with |
| 01:32.37 | ManxPower | RoyK, OpenPBX.org has taken the ONLY viable route to protest this -- fork the tree. |
| 01:33.12 | mog_work | yikes tzanger |
| 01:33.17 | RoyK | ManxPower: so it's ok for digium to continue abusing their position in the name of gpl? |
| 01:33.24 | ManxPower | puzzled, Honestly, I think the only thing that will heal the rift is for Digium to ONLY have a GPL license and stop ABE, and I don't see that happening anytime soon. |
| 01:33.27 | RoyK | you think that's a good thing? |
| 01:33.28 | tzanger | RoyK: unless you've got proof that's what they did, you're speculating |
| 01:33.40 | tzanger | http://www.mixdown.ca/~andrew/photos/KatieBirthday2005/img_4061.jpg |
| 01:33.42 | tzanger | that's my baby girl though |
| 01:33.45 | mog_work | manx that wouldnt heal the wound |
| 01:33.58 | tzanger | I have a feeling I'm gonna need to keep a shotgun by the door |
| 01:33.58 | mog_work | or at least not in my opinion |
| 01:34.00 | ManxPower | RoyK, You have taken the only viable route to fix that -- fork the codebase. |
| 01:34.11 | tzanger | ManxPower: that won't fix it |
| 01:34.16 | puzzled | ManxPower: that's why I asked about the Asterisk foundation :) aIt has been done before |
| 01:34.24 | puzzled | s/aIt/and it |
| 01:34.39 | tzanger | one of the biggest grips openpbx has with asterisk (and I agree with from time to time) is that Mark's not at this point willing to give up enough control to let it grow as fast as it needs |
| 01:34.39 | puzzled | is that the guy that shot the parent of the blond girl? |
| 01:34.41 | mog_work | that is not a bad idea puzzled |
| 01:34.46 | tzanger | puzzled: yeah |
| 01:34.56 | RoyK | tzanger: read that email or talk to bkw or both |
| 01:35.04 | RoyK | ManxPower: it's already forked |
| 01:35.05 | ManxPower | tzanger, Yes, I agree with that, but I'm not going and forking the code over it. |
| 01:35.06 | tzanger | RoyK: for the final time, I *READ* the email you posted |
| 01:35.09 | RoyK | ManxPower: get real |
| 01:35.10 | tzanger | it showed NOTHING |
| 01:35.15 | tzanger | RoyK: I've TALKED to BKW |
| 01:35.28 | ManxPower | tzanger, I guess if I was a programmer and not someone that just wants to get the job done for clients, I might care a bit more. |
| 01:35.31 | tzanger | as I said, Alison asked Digium and Digium said "it's your chocie" |
| 01:35.32 | RoyK | then start thinking |
| 01:35.33 | puzzled | seems to me we need some reflection and that excellent pakistan amdassador that leads the Tunis WSIS talks |
| 01:35.50 | tzanger | ManxPower: perhaps... but I don't agree a lot with RMSisms to begin with |
| 01:35.59 | ManxPower | tzanger, Me neither. |
| 01:36.01 | tzanger | he's done great things but he's also a fruit loop |
| 01:36.10 | tzanger | this fucking cat is sleeping and farting |
| 01:36.16 | RoyK | tzanger: as in "it's you choice. we will continue using you as our source of voice if you solemly stick to us........" |
| 01:36.21 | RoyK | or something? |
| 01:36.24 | tzanger | RoyK: no |
| 01:36.27 | tzanger | it's "It's your choice" |
| 01:36.27 | ManxPower | tzanger, I agree with that as well. RMS is a lunatic, but he has accomplished a lot of good stuff. |
| 01:36.34 | tzanger | there was none of that "we will continue using you.." bullshit |
| 01:36.38 | tzanger | that is the speculation on your part |
| 01:36.52 | tzanger | hell not even the email you posted said that |
| 01:36.56 | tzanger | all it said was "It's your choice" |
| 01:37.02 | RoyK | tzanger: so why would she suddenly stop serving other people? |
| 01:37.05 | tzanger | if you wish to speculate, then do so but CLEARLY INDICATE your speculation |
| 01:37.11 | ManxPower | tzanger, you and I both know the simple solution is for Mark to post a message making sure we hear HIS position on this issue. |
| 01:37.14 | asterboy | tdm400p clones would be nice, but what would be better is cheaper VOIP termination! |
| 01:37.16 | RoyK | she makes a living out of that for gods sake? |
| 01:37.20 | tzanger | RoyK: I don't know. But until you hear from her, you're speculating |
| 01:37.22 | drumkilla | RoyK: her own choice |
| 01:37.25 | drumkilla | RoyK: I know this for a fact. |
| 01:37.28 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
| 01:37.42 | tzanger | RoyK: as I said, she has made enough to BUY A HOUSE BEFORE DIGIUM CAME IN TO THE PICTURE... I seriously doubt digium's got tha tmuch influence over her |
| 01:37.46 | tzanger | seriously |
| 01:37.47 | mog_work | anyone at astricon who was there would know |
| 01:37.55 | RoyK | drumkilla: so suddenly she finds that all other projects except asterisk isn't worth working for? |
| 01:37.56 | tzanger | ManxPower: true, but Mark never does that |
| 01:38.21 | ManxPower | tzanger, It's one of his few Great Failings, in my opinion. |
| 01:38.23 | asterboy | I want $5 local VOIP termination and $15 unlimited NorthAmerican calling!!! |
| 01:38.30 | tzanger | ManxPower: agreed... he's extraordinarily busy |
| 01:38.31 | tzanger | and handsome |
| 01:38.34 | tzanger | oops |
| 01:38.36 | tzanger | I mean smart |
| 01:38.39 | puzzled | lol |
| 01:38.42 | RoyK | even if she can make just as much money from them as she can from * projects? |
| 01:38.44 | RoyK | strange........ |
| 01:38.45 | puzzled | asterboy: start your own clec |
| 01:38.45 | ManxPower | asterboy, Yes, and I want an 8-ball of coke, but we can't always get what we want. |
| 01:38.48 | Math` | astcryz: uhm vbuzzer has 8$us unlimited DOD to north america |
| 01:38.55 | asterboy | lol |
| 01:39.07 | puzzled | ManxPower: dutch coke or the other kind? |
| 01:39.10 | asterboy | Math: sweet! |
| 01:39.25 | ManxPower | puzzled, I was joking. Coke is far overrated. |
| 01:39.39 | puzzled | indeed it is |
| 01:39.48 | Math` | asterboy: they say you have to use their software, but its SIP with plaintext auth :) |
| 01:39.57 | tzanger | RoyK: *sigh* ask her |
| 01:40.06 | puzzled | RoyK: does she work for free for Digium? |
| 01:40.11 | tzanger | but please for the love of god please stop whining about it here. We dont' have enough information |
| 01:40.22 | mog_work | no puzzled |
| 01:40.25 | kn0x | manxpower- back |
| 01:40.43 | kn0x | WARNING: /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko needs unknown symbol rtc_register WARNING: /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko needs unknown symbol rtc_unregister WARNING: /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko needs unknown symbol rtc_control |
| 01:40.51 | puzzled | mog_work: it was rhetorical :) at least I assumed the relation was commercial |
| 01:40.55 | RoyK | puzzled: no. i guess not. but i guess she's been told to keep away from the "voip terrorists" |
| 01:40.56 | tzanger | kn0x: you need rtc support in your kenrel |
| 01:40.58 | tzanger | er kernel |
| 01:40.59 | mog_work | load rtc or crc kn0x |
| 01:41.03 | mog_work | i cant ever remeber |
| 01:41.06 | ManxPower | It seems to me all the high string people went to OpenPBX. Maybe it should be called EctomorphPBX? |
| 01:41.09 | kn0x | built in? |
| 01:41.09 | tzanger | rtc_register |
| 01:41.11 | tzanger | that's RTC |
| 01:41.17 | mog_work | well some people are crazy puzzled... |
| 01:41.22 | tzanger | Ectomorph? |
| 01:41.33 | kn0x | here i have it installed as a module |
| 01:41.34 | ManxPower | tzanger, I think is problem is that he compiled RTC as part of the kernel and not as a module. |
| 01:41.41 | kn0x | WARNING: /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko needs unknown symbol rtc_register WARNING: /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko needs unknown symbol rtc_unregister WARNING: /lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko needs unknown symbol rtc_control |
| 01:41.42 | mog_work | naj |
| 01:41.43 | tzanger | yeah |
| 01:41.43 | ManxPower | tzanger, "high strung" |
| 01:41.48 | mog_work | he doesnt have module loaded |
| 01:41.50 | tzanger | kn0x: stop repeating |
| 01:41.51 | puzzled | RoyK: have you been sniffing some norwegian strawberries again? |
| 01:42.10 | kn0x | manx told me to remove it and add as a module |
| 01:42.21 | kn0x | thats done but now it still has a few missing symbols |
| 01:42.33 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-327da59287f905f0) |
| 01:42.46 | ManxPower | kn0x, does the Zaptel readme or docs have anything useful to say? |
| 01:42.48 | puzzled | so anyone up for a cold beer? |
| 01:42.51 | tzanger | kn0x: what's modinfo say for ztdummy |
| 01:43.31 | RoyK | puzzled: they're good for eating, but not at this time of the year |
| 01:43.40 | ManxPower | puzzled, if it's dark, I'll take it |
| 01:43.43 | puzzled | RoyK: ok :) |
| 01:43.44 | asterboy | There should be a way to build an interface to PC softphones by hooking up the sound card to linux sound card and some sort of script to control the interface. |
| 01:44.02 | ManxPower | asterboy, All softphones suck. |
| 01:44.05 | puzzled | ManxPower: when did the British get to you?! |
| 01:44.06 | asterboy | Do that and you can route calls. |
| 01:44.07 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 01:44.09 | hhoffman | mmmmm porter :-~ |
| 01:44.22 | ManxPower | puzzled, I never thought I liked beer -- until I tried Guinness |
| 01:44.32 | asterboy | Yes, thats the point...softphone turned into a hardphone. |
| 01:44.32 | kn0x | about 1/8 of the number of original missing symbols |
| 01:44.34 | drumkilla | ugh, Guinness sucks |
| 01:44.39 | drumkilla | RoyK: it's probably backed up |
| 01:44.41 | drumkilla | give it some time |
| 01:44.48 | ManxPower | drumkilla, only guinness in a bottle |
| 01:44.50 | asterboy | Guinness rocks! |
| 01:44.50 | kn0x | modinfo ztdummy parmtype: debug:int description: Dummy Zaptel Driver author: Robert Pleh <robert.pleh@hermes.si> license: GPL vermagic: 2.6.13-gentoo-r5 SMP preempt PENTIUMIII gcc-3.3 depends: zaptel |
| 01:44.58 | asterboy | Steak and eggs in a glass! |
| 01:45.02 | puzzled | ManxPower: are you sure you were naturally born and not genetically modified by the dark side? |
| 01:45.07 | tzanger | hmm that's it? |
| 01:45.13 | tzanger | modprobe zaptel |
| 01:45.15 | tzanger | make sure it loads |
| 01:45.19 | tzanger | modprobe ztdummy |
| 01:45.24 | ManxPower | tzanger, he doesn't listen very well. |
| 01:45.32 | ManxPower | puzzled, I'm pretty sure. |
| 01:45.33 | hhoffman | guinness needs to be in the can and not the bottle though |
| 01:45.44 | ManxPower | hhoffman, I totally agree with that. |
| 01:45.44 | tzanger | guinness is far too heavy for me |
| 01:45.53 | puzzled | so many people so many flavors |
| 01:45.59 | kn0x | modinfo ztdummy parmtype: debug:int description: Dummy Zaptel Driver author: Robert Pleh <robert.pleh@hermes.si> license: GPL vermagic: 2.6.13-gentoo-r5 SMP preempt PENTIUMIII gcc-3.3 depends: zaptel |
| 01:46.01 | asterboy | its only good at the beginning when the widget has exploded. |
| 01:46.02 | ManxPower | Well, it needs to be in a glass and stored in a keg, but if you can't do that, guinness in a can is decent |
| 01:46.07 | Math` | lol |
| 01:46.07 | kn0x | thats the modinfo tzanger |
| 01:46.22 | Math` | (for the norwegian strawberries of earlier, damn scroll) |
| 01:46.27 | konfuzed | finally the amd athlon 64 2500+ arrived |
| 01:46.38 | puzzled | Math`: :) |
| 01:46.44 | konfuzed | something more worthwhile to install asterisk on |
| 01:47.07 | puzzled | konfuzed: I had fun moving all stuff from /usr/lib to /usr/lib64 |
| 01:47.10 | konfuzed | puzzled: you shold try chocolate stout |
| 01:47.13 | *** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org) |
| 01:47.22 | puzzled | konfuzed: isn't that a girly drink? |
| 01:47.23 | hhoffman | is there a way to get custom words from the attendents voice? |
| 01:47.37 | konfuzed | cacao is way under rated and misunderstood |
| 01:47.43 | asterboy | Are there services that don't require the asterisk or pbx and connects you directly via Internet...Basically use an IP phone direct connected. |
| 01:47.44 | asterboy | ? |
| 01:47.46 | ManxPower | hhoffman, thevoice.digium.com or voice.digium.com |
| 01:47.59 | hhoffman | ManxPower: oh! thanks |
| 01:48.07 | ManxPower | asterboy, Yes, almost every ITSP out there. |
| 01:48.07 | tzanger | asterboy: yes, many |
| 01:48.15 | konfuzed | puzzled: chocolate fondue ladies is even better |
| 01:48.29 | konfuzed | there s all kind s of important uses for cacao |
| 01:48.31 | konfuzed | ;^) |
| 01:48.43 | asterboy | makes sense since you can do with with offices across the country. |
| 01:48.44 | puzzled | konfuzed: better get to them before the are covered in it. 9.5 weeks doesn't realy work in reality |
| 01:48.53 | hhoffman | there are some funny pre-recorded voices here "carried-away-by-monkeys.gsm" |
| 01:48.56 | konfuzed | my asterisk on debian is hurting me though |
| 01:49.05 | tzanger | konfuzed: did you use the asterisk .deb? |
| 01:49.07 | puzzled | seems appropriate then |
| 01:49.16 | konfuzed | tzanger: sort of |
| 01:49.17 | denon | why use the .dep? |
| 01:49.19 | tzanger | haha |
| 01:49.19 | denon | er .deb |
| 01:49.26 | tzanger | I don't recommend the .deb I was just asking |
| 01:49.28 | puzzled | why use debian? |
| 01:49.33 | denon | debian itself is fine |
| 01:49.35 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 01:49.36 | konfuzed | I found out there are good reasons too |
| 01:49.43 | denon | but asterisk and packages dont seem to make sense together |
| 01:49.49 | denon | asterisk's development cycle is too rapid |
| 01:49.52 | puzzled | there are no reasons, only preferences |
| 01:50.17 | puzzled | as we say in Dutch: what the farmer does not like he will not eat |
| 01:50.34 | asterboy | lol, this industry is turning into a Watkins via its multi-tier marketing. |
| 01:50.38 | konfuzed | the asterisk package in debian actually has ast-1.2.0.rc1 in there |
| 01:50.45 | konfuzed | I should say available |
| 01:50.48 | denon | except RC2 is out |
| 01:50.49 | konfuzed | not by default |
| 01:50.56 | puzzled | konfuzed: that's old stuff. get rc2 |
| 01:50.59 | konfuzed | by default it is stable of course |
| 01:51.08 | puzzled | 5 year old stable :) |
| 01:51.15 | denon | it's still asterisk, isnt it? |
| 01:51.27 | denon | I'm not sure the packager can promise stability :) |
| 01:51.44 | asterboy | interesting...any of those ITSPs have a fixed plan? |
| 01:51.50 | konfuzed | i spoke too soon |
| 01:51.52 | puzzled | asterboy: call them |
| 01:51.55 | konfuzed | Experimental 1:1.2.0-rc2.dfsg-1 |
| 01:51.59 | ManxPower | puzzled, Considering dutch cuisine...... |
| 01:52.05 | asterboy | there is a lot to investigate. |
| 01:52.18 | ManxPower | But at least the Dutch have awesome candy and cookies and pretty decent beer. |
| 01:52.29 | puzzled | :) |
| 01:52.31 | konfuzed | any way ive butted my head too long on this one |
| 01:52.41 | ManxPower | puzzled, I'm ordering some via mailorder as soon as my finances get back on track |
| 01:52.43 | denon | and windmills |
| 01:52.46 | denon | and wooden shoes! |
| 01:53.00 | denon | ..and the red light district |
| 01:53.01 | konfuzed | if i cant get past this current features.so error (after dinner) then Im gonna start over from the install |
| 01:53.09 | puzzled | and our crap cheese we export to the rest of the world and make them believe it's the best |
| 01:53.20 | denon | puzzled: we all know better . . |
| 01:53.25 | puzzled | ManxPower: shoot me an email and I'll send you some |
| 01:53.34 | YoMama | so...what's the cheapest ATA? |
| 01:53.39 | puzzled | denon: that's definitely not dutch local cheese |
| 01:53.40 | ManxPower | puzzled, I came back from the market in Amstersdam while I was there and set the bag of stuff on the counter, and the stroopwaffels bag fell out and the counter person said "Ah, I see you discovered stroopwaffels." |
| 01:53.41 | denon | YoMama: dunno, grandstreams? |
| 01:53.46 | denon | puzzled: *nod* |
| 01:53.46 | docelm0 | «YoMama» Linksys |
| 01:53.52 | puzzled | ManxPower: nice :) |
| 01:53.53 | tzanger | wow |
| 01:53.57 | tzanger | nanoblogger looks nice! |
| 01:54.13 | YoMama | docelm0: what about those locked sipuras that can be unlocked? |
| 01:54.31 | docelm0 | I dont know.. I buy them unlocked.. :) |
| 01:54.37 | puzzled | there shall be no illegal unlocking in this chan |
| 01:54.40 | asterboy | gotta start a site like distro watch listing all ther services indexed by price of course |
| 01:55.06 | ManxPower | I ate the entire package on my way to (what's that city with the Phillips Museum in it?) |
| 01:55.07 | YoMama | puzzled: and it's illegal? |
| 01:55.14 | puzzled | asterboy: check the asterisk users list archives. zoa recently announced such a site |
| 01:55.40 | puzzled | YoMama: in the US I guess the DMCA comes into play. but IANAL. |
| 01:55.47 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-c379a7235e93a528) |
| 01:55.53 | YoMama | DMCA? |
| 01:55.59 | puzzled | YoMama: why not buy the real thing? |
| 01:56.08 | ManxPower | puzzled, when I get moved into a perm place I'll send you a paypal for the cost and you can ship me a case of them |
| 01:56.24 | puzzled | ManxPower: of stroopwafels? |
| 01:56.30 | ManxPower | puzzled, *nod* |
| 01:56.32 | tzanger | what the hell's a stroopwafel? |
| 01:56.38 | puzzled | man that's a lot of stroopwaffels |
| 01:56.51 | ManxPower | tzanger, it's a totally awesome wafflecookie thing with carmel in it. |
| 01:56.59 | tzanger | sounds nice |
| 01:57.04 | ManxPower | puzzled, I'd give them to friends and freeze the rest. |
| 01:57.07 | YoMama | puzzled: 'cause it's cheaper maybe? |
| 01:57.07 | tzanger | I'm used to most dutch things involving fish |
| 01:57.08 | puzzled | tzanger: come to .nl and enjoy one of the few legal things :) |
| 01:57.15 | tzanger | puzzled: :-) |
| 01:57.18 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 01:57.20 | puzzled | YoMama: no because they are not locked |
| 01:57.25 | *** join/#asterisk xiel (n=xiel@pcp08477368pcs.summit01.tn.comcast.net) |
| 01:57.31 | puzzled | tzanger: herring! |
| 01:57.41 | ManxPower | puzzled, I had a lot of....fun...while I was there. |
| 01:57.47 | puzzled | hehehe |
| 01:58.07 | klydal | im going to .nl next month |
| 01:58.08 | kn0x | im sorry manxpower... im back for now |
| 01:58.14 | klydal | only going to be there for 4 hours though :( |
| 01:58.17 | kn0x | * for real |
| 01:58.19 | ManxPower | klydal, the 'shrooms will kick your ass. |
| 01:58.23 | puzzled | ManxPower: never thought about freezing them. usually just eat the package |
| 01:58.27 | kn0x | any other sugestions |
| 01:58.31 | klydal | they sell themn at the airport? |
| 01:58.44 | ManxPower | klydal, no, you need to be there for at least a day |
| 01:58.50 | puzzled | klydal: sure just go to a sweets shop and ask for "stroopwafels" |
| 01:58.56 | klydal | they have that speed train that goes to red light district |
| 01:59.07 | ManxPower | puzzled, those are much more addictive than 'shrooms! |
| 01:59.16 | puzzled | :) |
| 01:59.18 | klydal | stroopwafel? ok I will try |
| 01:59.28 | ManxPower | klydal, Actually to the main station, just outside the red light discrict |
| 01:59.40 | puzzled | ManxPower: iirc they tried to make those illegal (haha) but they failed (more haha) |
| 01:59.49 | ManxPower | puzzled, 'scrooms? |
| 01:59.54 | ManxPower | ..er.. 'shrooms? |
| 02:00.00 | kn0x | its rtc is still missing is still missing some keywords |
| 02:00.07 | puzzled | ManxPower: abbreviation for mushrooms |
| 02:00.18 | ManxPower | puzzled, yes, I know. |
| 02:00.22 | kn0x | *symbols, not keyword |
| 02:00.24 | Math` | common psylocibin |
| 02:00.40 | puzzled | sounds like a prefectly legal substance here |
| 02:00.47 | Math` | really? |
| 02:01.05 | klydal | I would probably rather partake of the female services. I have had my share of trips |
| 02:01.06 | puzzled | most likely and if not they won't jail you if its only for personal use |
| 02:01.11 | ManxPower | puzzled, I didn't even know they are legal there, but once I found out I figured "Why not? I'm pretty experienced with this stuff!". I bought the middle strength and took half of a dose. Gawd! Knocked me on my ass. REALLY glad I was in the hotel room. |
| 02:01.12 | kn0x | haha |
| 02:01.46 | ManxPower | Dutch 'shrooms humbled me. |
| 02:01.57 | Math` | I only tried em once |
| 02:01.58 | klydal | better than the home made 8balls? |
| 02:02.02 | puzzled | ManxPower: they are dangerous indeed. many people thought they could fly while being on the fourth floot |
| 02:02.09 | Math` | lolllll |
| 02:02.27 | Math` | funny but sad |
| 02:02.34 | ManxPower | puzzled, only are not careful or unstable in the first place. |
| 02:02.46 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 02:02.46 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 02:03.01 | Qwell | Anybody happen to be using realtime static and realtime (proper) at the same time, with the same configs? The wiki says it isn't possible, but I'm not believing it |
| 02:03.59 | konfuzed | ive been trying to find a cause for this bot so far no go |
| 02:04.04 | konfuzed | [chan_features.so]Nov 14 21:05:48 WARNING[6871]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_features.so: undefined symbol: ast_register_file_version |
| 02:04.04 | konfuzed | Nov 14 21:05:48 WARNING[6871]: loader.c:440 load_modules: Loading module chan_features.so failed! |
| 02:04.18 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 02:04.22 | konfuzed | thats what I get when I run asterisk -cvvvvvvvvv |
| 02:04.54 | konfuzed | how do I fix that or turn it off |
| 02:05.02 | tzanger | there is a feature *channel* driver? |
| 02:05.25 | puzzled | isn't that the #9 stuff? |
| 02:05.37 | puzzled | plays the monkey thing while on a call |
| 02:05.56 | kn0x | manxpower- what should i do now |
| 02:06.09 | konfuzed | is feature *channel* driver required or can asterisk work with out it |
| 02:06.15 | konfuzed | sounds pretty central |
| 02:06.19 | kn0x | i changed rtc from part of the kernel to a module |
| 02:06.21 | puzzled | RoyK: so long for the conspirancy theory :) |
| 02:06.50 | kn0x | manxpower- its still giving me those four missing symbols |
| 02:06.54 | kn0x | less but still missing |
| 02:07.31 | ManxPower | kn0x, I have no more suggestions other than to check the mailinglist archives |
| 02:07.33 | tzanger | I know of res_features but chan_features?! |
| 02:07.34 | ManxPower | ~mailinglist |
| 02:07.36 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 02:08.30 | tzanger | :-) |
| 02:08.35 | tzanger | I'd love to try .nl shrooms |
| 02:08.41 | tzanger | but from what you describe I best have a spotter |
| 02:08.45 | tzanger | or at least someone with a videocamera |
| 02:08.53 | konfuzed | puzzled: for my shrimp mushroom and seeweed soup |
| 02:09.29 | ManxPower | tzanger, a sober guide is usually best |
| 02:09.34 | konfuzed | ok so where can I turn of the feature *channel* to find out for sure |
| 02:10.01 | *** part/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:10.05 | RoyK | puzzled: wot? |
| 02:10.15 | puzzled | RoyK: your emails came through |
| 02:10.32 | RoyK | the one to -dev? |
| 02:10.34 | puzzled | konfuzed: /etc/asterisk/modules.conf |
| 02:10.37 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:10.37 | RoyK | or the ones to -user? |
| 02:10.50 | puzzled | -dv |
| 02:10.52 | puzzled | -dev |
| 02:10.56 | klydal | wow, I thought asterisk was like a softphone or something. this stuff is way over my head :P |
| 02:11.05 | harryvv | iptel.org is down? |
| 02:11.14 | RoyK | puzzled: it's the ones to -users that's gone missing |
| 02:11.15 | puzzled | klydal: have some shrooms and all will be clear |
| 02:11.19 | harryvv | flydal, no its a server |
| 02:11.30 | ManxPower | klydal, all softphones suck. Asterisk doesn't suck. Ergo Asterisk is not a softphone. |
| 02:11.42 | puzzled | lol |
| 02:11.50 | klydal | :) sorry, I found this through a link from my voip provider |
| 02:12.07 | sivana | anyone here offering local voip service in Canada? |
| 02:12.09 | RoyK | puzzled: could you try to send a ping to users, plese? |
| 02:12.10 | harryvv | what are you trying to do |
| 02:12.11 | puzzled | RoyK: what time did you post it? |
| 02:12.18 | harryvv | sibana, where are you at |
| 02:12.29 | RoyK | From: asterisk-users-bounces@lists.digium.com |
| 02:12.29 | RoyK | Subject: Your message to Asterisk-Users awaits moderator approval |
| 02:12.29 | RoyK | Date: 15. november 2005 02.05.41 GMT+01:00 |
| 02:12.30 | harryvv | sivana what province |
| 02:12.36 | puzzled | RoyK: I have emails coming in at (most recent) 02.49am |
| 02:12.52 | sivana | I'm in Ontario.. but I want to copy someone's 911 notification as per CRTC Telecom Decision 2005-61 |
| 02:12.55 | sivana | :) |
| 02:12.59 | puzzled | RoyK: maybe you sent it from the wrong account? |
| 02:13.01 | RoyK | i've had none since 0157 CET |
| 02:13.04 | RoyK | no way |
| 02:13.21 | RoyK | it's sent to the account on the list |
| 02:13.27 | sivana | I'm trying to write my own, but I realize that I suck :) |
| 02:13.27 | puzzled | RoyK: kick your silly exchange server |
| 02:13.34 | konfuzed | there is no chan_feature of any kind in modules.conf |
| 02:13.38 | konfuzed | not much in there really |
| 02:13.42 | RoyK | puzzled: postfix+cyrus imap |
| 02:13.52 | konfuzed | most of it has been set to noload |
| 02:14.05 | Math` | sivana: do you have PRIs? |
| 02:14.11 | puzzled | RoyK: there you go. you should know better than to use a mailserver written by a dutch guy |
| 02:14.11 | sivana | yes |
| 02:14.20 | harryvv | sivana what do the pris cost you |
| 02:14.22 | Math` | sivana: whats the pricing? |
| 02:14.30 | konfuzed | when I run asterisk -cvvvvvvvvv |
| 02:14.32 | Math` | Im in quebec, must be around the same prices |
| 02:14.34 | sivana | $850/mo |
| 02:14.38 | Math` | for a T1? |
| 02:14.39 | puzzled | ouch |
| 02:14.42 | RoyK | puzzled: strange thing i get messages from -dev but not -users |
| 02:14.42 | Math` | $cad? |
| 02:14.43 | harryvv | for one pri? t-1? |
| 02:14.46 | sivana | yup |
| 02:14.46 | RoyK | straaaaaaaaange |
| 02:14.47 | sivana | PRI |
| 02:15.00 | sivana | $700 + 150 local loop fee |
| 02:15.00 | Math` | sivana: how much per DID? |
| 02:15.00 | puzzled | RoyK: maybe they unsubscribed you :) |
| 02:15.04 | sivana | $2 |
| 02:15.10 | ManxPower | Um, $850/mmonth for a 23B+1D channel PRI is NOT all that expensive |
| 02:15.13 | harryvv | yea, thats about right. here in bc one pri can be from 650-1,100 dollars per month. |
| 02:15.18 | Math` | sivana: do you pay for local outbound or inbound? |
| 02:15.24 | sivana | no |
| 02:15.30 | sivana | unlimited local in/out |
| 02:15.41 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
| 02:15.44 | Math` | sivana: whats your provider? |
| 02:15.48 | sivana | lol |
| 02:15.49 | puzzled | but who calls local these days |
| 02:16.02 | Math` | lol |
| 02:16.03 | sivana | we've got 70+ residential customers... so lots |
| 02:16.09 | Math` | hehe |
| 02:16.38 | RoyK | puzzled: nope. subscribed..... |
| 02:16.40 | puzzled | sivana: tell em to move abroad. better for business |
| 02:16.45 | sivana | :) |
| 02:17.12 | puzzled | RoyK: digest turned on? sony bmg rootkit installed on your exchange server? :) |
| 02:17.32 | Math` | puzzled: the DRM stuff? |
| 02:17.36 | puzzled | yup |
| 02:17.37 | harryvv | sivana, who is your provider of pri |
| 02:17.39 | puzzled | evil stuff |
| 02:17.47 | Math` | puzzled: yeah I read the slashdot article |
| 02:17.49 | sivana | you guys are away that any CDN local voip provider must send out 911 limitations/explanation notifications |
| 02:17.53 | Math` | ManxPower: lol |
| 02:18.00 | sivana | aware |
| 02:18.07 | sivana | harryvv: a local CLEC :) |
| 02:18.08 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 02:18.08 | puzzled | ManxPower: if you put me on it you will not get your stroopwafels |
| 02:18.10 | harryvv | sivana explain |
| 02:18.13 | sivana | you need a PRI? |
| 02:18.18 | harryvv | who is the local clec? |
| 02:18.19 | sivana | http://www.crtc.gc.ca/archive/ENG/Decisions/2005/dt2005-61.htm |
| 02:18.20 | ManxPower | puzzled, you are not paranoid |
| 02:18.22 | konfuzed | undefined symbol: ast_register_file_version how do I fix this or find the source of this problem |
| 02:18.24 | sivana | ExaTel |
| 02:18.28 | puzzled | ManxPower: phew :) |
| 02:18.41 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 02:18.50 | harryvv | exatel mmm interesting. thay own the infra? |
| 02:19.08 | RoyK | puzzled: strange thing. all of a sudden delivery was turned off |
| 02:19.10 | sivana | their own.. I own mine |
| 02:19.19 | Math` | how do you actually route 911/E911 |
| 02:19.23 | puzzled | konfuzed: seems it is not defined so include asterisk.h or something (I am *no* C coder) |
| 02:19.28 | sivana | they only provide me with the PRI.. we have our own local circuits |
| 02:19.35 | harryvv | sibana, okay thats cool. |
| 02:19.56 | puzzled | RoyK: hooray. that will be 1 norwegian DID please |
| 02:20.14 | sivana | Math`: we do it in-house... |
| 02:20.20 | RoyK | puzzled: but still my emails returns from a moderator |
| 02:20.41 | Math` | sivana: you mean you have your 911 operators at your office and you just call the police manually? :P |
| 02:20.50 | sivana | ok... so I take it you guys aren't compliant and have no notification I can steal :) |
| 02:20.53 | RoyK | puzzled: so no DIDs for you, mate |
| 02:20.57 | puzzled | RoyK: afaik it only returns those if the email address has changed. maybe someone guessed your password and changed it to billg@microsoft.com |
| 02:20.57 | harryvv | sivana, what services does your company sell |
| 02:21.19 | puzzled | RoyK: half a DID then? |
| 02:21.19 | sivana | Math`: we have our own operators.. soon to launch http://www.911route.com |
| 02:21.26 | konfuzed | shite /usr/src/asterisk/ does not have asterisk.h where do I find it |
| 02:21.32 | sivana | harryvv: local voip |
| 02:21.40 | RoyK | puzzled: not really likely |
| 02:21.43 | puzzled | konfuzed: /usr/src/asterisk/include/asterisk.h |
| 02:22.00 | puzzled | RoyK: then no shrooms for you! |
| 02:22.01 | RoyK | puzzled: it just matches up with the time i was trolling about that case |
| 02:22.17 | RoyK | puzzled: fucking imperialistic idiots |
| 02:22.50 | Math` | is there any document describing how telcos are interconnected? (tier stuff, signalling etc...) |
| 02:22.55 | puzzled | RoyK: settle down and just subscribe using a different address. send email again and see what happens |
| 02:23.01 | supaigtr | Math: SS7 and ISUP |
| 02:23.08 | harryvv | sivana, how is your company dealing with the new rates from telus and now rogers offering voip? |
| 02:23.20 | puzzled | Math`: SS7, h323 and sip |
| 02:23.28 | Math` | supaigtr: I knew about ss7... but, is there any authorities such as ARIN for ip addresses and ASN |
| 02:23.31 | sivana | harryvv: no competition... they can't compete here |
| 02:23.40 | sivana | harryvv: www.voctel.com |
| 02:23.44 | harryvv | ohh why not thay dont have service there? |
| 02:23.47 | supaigtr | Huh? U mean IP? |
| 02:23.49 | sivana | nope :) |
| 02:24.09 | sivana | our only major competitor is Bell and Cogeco |
| 02:24.14 | harryvv | well thats not going to last. so basisly you are in a market thay are not interested? |
| 02:24.17 | sivana | both of which are priced outside the local market |
| 02:24.22 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 02:24.25 | puzzled | Math`: afaik there are no IP addresses involved in SS7. you choose your own on the backside |
| 02:24.42 | sivana | harryvv: we have 54k population in northern Ontario |
| 02:24.48 | Math` | puzzled: I know SS7 isnt related to IP, but how is routing performed using plain telephony systems |
| 02:24.52 | Math` | lets say internet doesnt exist |
| 02:25.09 | puzzled | Math`: SS7 and tdm (E1...Oc768) links |
| 02:25.13 | klydal | so whats an advantage of having a pbx for yourself? |
| 02:25.19 | IronHelix | flexibility |
| 02:25.24 | IronHelix | and scalability |
| 02:25.25 | puzzled | klydal: leetness |
| 02:25.26 | Math` | puzzled: is there any authority that manages links? |
| 02:25.29 | LostFrog | lol |
| 02:25.30 | IronHelix | that too |
| 02:25.33 | LostFrog | 1337. |
| 02:25.42 | supaigtr | Math: Its ISUP and ss7. ss7 has its own addressing. |
| 02:25.42 | IronHelix | with your own pbx you can have as many internal extensions as you want |
| 02:25.47 | IronHelix | without paying extra telco costs |
| 02:25.48 | ManxPower | klydal, what to know what the advantage of having my own PBX is? |
| 02:25.53 | ManxPower | ..er... want |
| 02:25.57 | harryvv | sivana thats good. so obviosly the big players are not interested in smaller towns. |
| 02:26.05 | klydal | sure manxpower |
| 02:26.06 | puzzled | Math`: it's decided between the 2 parties that interconncet. usually the regulator hands out the pointcodes (sort of IP addresses) |
| 02:26.09 | sivana | ya |
| 02:26.10 | LostFrog | Flexibiliy and scalability. |
| 02:26.13 | supaigtr | Math: Its not that simple. You have to get with an exsising SS7 provider. Need at least 2 A-Links 56k.] |
| 02:26.26 | ManxPower | klydal, It lets me prototype features, options, and upgrades for the Asterisk systems I install for my corporate clients 8-) |
| 02:26.45 | klydal | yeah thats cool |
| 02:26.47 | ManxPower | supaigtr, you are in the USA/CA, right? |
| 02:26.53 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 02:26.55 | klydal | knowledge is definitely power |
| 02:26.59 | konfuzed | ok so asterisk.h has a bunch of entries mostly with int and some with void - int seems like defining a variable but what about void? is that essentially diabling that particular variable or function? |
| 02:27.01 | puzzled | ManxPower: "prototype" nice one :) |
| 02:27.32 | LostFrog | void is usually means one of two things.. |
| 02:27.40 | IronHelix | at its most basic, * can be little more than a cool answering machine on a voip line. in more advanced configurations, you can run 100's of extensions and multiple locations |
| 02:27.47 | konfuzed | klydal: I'm proof that just a little powerful knowledge can be dangerous |
| 02:27.52 | konfuzed | mostly to myself of course |
| 02:27.54 | konfuzed | ;^) |
| 02:27.54 | LostFrog | A function does return anything or a pointer that can hold lots of different things. |
| 02:28.02 | LostFrog | that does not. |
| 02:28.17 | ManxPower | In the USA/CA it takes lots of money, lawyers, and patience to get an SS7 connection into the telco. In may other parts of the world the telco wants to hand you a ss7 connection instead of a PRI |
| 02:28.26 | puzzled | konfuzed: look at app_skeleton.c or at a app_<some_module>.c to see where all stuff gets defined and which includes it uses |
| 02:29.03 | supaigtr | SS7 and switch in the US = have at least a few million to get everything up and working. |
| 02:29.03 | puzzled | ManxPower: except Level3, MCI and soon Telia |
| 02:29.13 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 02:29.15 | puzzled | ManxPower: that is, they allow SIP too |
| 02:29.34 | supaigtr | How do them term faxes over SIP interconnections? |
| 02:29.39 | konfuzed | the items that asterisk -cvvvvvvv xomplains about is lsited as void in asterisk.h |
| 02:29.56 | puzzled | supaigtr: read about T.38 |
| 02:29.58 | konfuzed | id like to change it to int instead of void but that sounds just a little too easy |
| 02:30.18 | supaigtr | Yea. T.38 can = unrealiable. |
| 02:30.19 | konfuzed | would that work? |
| 02:30.49 | Flauto | is there anyone using vbuzzer with asterisk? |
| 02:30.56 | drumkilla | konfuzed: I have no idea what you are talking about, but I can guarantee that will not work |
| 02:31.00 | drumkilla | :) |
| 02:31.05 | LostFrog | konfuzed: give me one example. |
| 02:31.10 | puzzled | konfuzed: no idea, I'll stick to serving the channel with cold beers |
| 02:31.45 | konfuzed | == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) |
| 02:31.45 | konfuzed | <PROTECTED> |
| 02:31.45 | konfuzed | Nov 14 21:19:57 WARNING[6903]: loader.c:440 load_modules: Loading module chan_features.so failed! |
| 02:32.22 | LostFrog | It's a function that returns nothing. |
| 02:32.34 | konfuzed | music on hold has a chan_res entry in modules.conf I believe |
| 02:32.39 | konfuzed | could that be the problem |
| 02:32.39 | puzzled | since I think it's called res_features there seems to be more basic things wrong |
| 02:33.00 | konfuzed | LostFrog: do you mean my error |
| 02:33.07 | RoyK | fucking morons |
| 02:33.19 | IronHelix | ? |
| 02:33.36 | puzzled | RoyK: temper, temper |
| 02:33.38 | Flauto | nobody uses vbuzzer? at least canadians should know it |
| 02:33.38 | RoyK | banning people from a mailing list just because they speak out is bad, bad, bad |
| 02:34.00 | Flauto | royk what happened |
| 02:34.01 | LostFrog | What version of * are you using, konfuzed? |
| 02:34.17 | IronHelix | seems it would be difficult to effectively ban somebody from a public list, why couldnt they just sign up again with a fresh hotmail or something |
| 02:34.25 | RoyK | Flauto: read -dev |
| 02:34.42 | Flauto | change and email address |
| 02:34.46 | Flauto | you will be in again |
| 02:34.54 | sivana | hehe |
| 02:34.55 | RoyK | Flauto: not the point |
| 02:35.05 | puzzled | RoyK: you mean the "Det virker!" response which eludes me |
| 02:35.07 | LostFrog | That is a weird error. |
| 02:35.08 | konfuzed | LostFrog: hhmmm its changed ( on this box ) I thinks its 1.0.7 |
| 02:35.08 | RoyK | Flauto: removing me from the list is BAD |
| 02:35.13 | konfuzed | how do I confirm that again |
| 02:35.22 | puzzled | konfuzed: use 1.2.0-rc2 |
| 02:35.25 | konfuzed | yeah |
| 02:35.28 | konfuzed | 1.07 |
| 02:35.30 | drumkilla | konfuzed: you have a module from 1.2 on a 1.0 asterisk core |
| 02:35.30 | RoyK | puzzled: that was on -dev. "det virker" means "it works" |
| 02:35.37 | drumkilla | konfuzed: rm -rf /usr/lib/asterisk/modules |
| 02:35.39 | drumkilla | then install it again |
| 02:35.42 | konfuzed | that;s very plausible |
| 02:35.42 | LostFrog | That's what it looks like, RoyK. |
| 02:35.43 | drumkilla | 'then you will be good to go |
| 02:36.08 | konfuzed | this install has been bugging me so now Ive made the boo boo of back and forth a little |
| 02:36.16 | RoyK | puzzled: but on -users i get no messages and cannot post |
| 02:36.19 | Flauto | well, royk, i have been booted out from this room many times because i was asking stupide questions, well, at least people here thing they were stupid questions. but i still need to come back to ask, that is how i can learn. i ask morre and i learn more |
| 02:36.39 | konfuzed | i was already ready to reinstall it again |
| 02:36.41 | Qwell | Flauto: No such thing as stupid questions |
| 02:36.47 | Qwell | (just stupid people) |
| 02:36.59 | puzzled | Flauto: buy the book and become enlightened (partially at least) |
| 02:37.05 | konfuzed | just malformed questions |
| 02:37.16 | RoyK | Flauto: this was for blaming digium for acting monopoistic. and it was on a figgin list, not on irc |
| 02:37.17 | LostFrog | Qwell: how can I banish Bill Gates legally? |
| 02:37.47 | puzzled | LostFrog: go to eastern europe and hire someone |
| 02:37.49 | hhoffman | anyone have a favorite soft-phone client for use under linux? thinking about giving mozphone a whirl |
| 02:38.32 | LostFrog | puzzled: I already did.. they threw pie at him. :) |
| 02:38.32 | puzzled | hhoffman: xten lite, sjphone, and others which can be found on voip-info.org |
| 02:38.32 | LostFrog | snom 360 |
| 02:38.32 | puzzled | LostFrog: yeah that one was fun |
| 02:38.43 | klydal | anyone of any suggestions for voip providers? |
| 02:38.54 | klydal | *have |
| 02:38.54 | Flauto | puzzled, when you just start out, it is hard enough to even find a book to read |
| 02:38.56 | puzzled | klydal: find them at voip-info.org |
| 02:39.04 | IronHelix | if you want lines, try quantumvoice or broadvoice |
| 02:39.05 | klydal | oooo, thank yous |
| 02:39.05 | puzzled | Flauto: ateriskdocs.org |
| 02:39.06 | LostFrog | voicepulse, broadvoice, RTFW? |
| 02:39.08 | hhoffman | I'm using teliax and have had really good luck |
| 02:39.14 | puzzled | Flauto: make that asteriskdocs.org |
| 02:39.18 | IronHelix | if you want minutes, try voicepulse or nuphone |
| 02:39.29 | IronHelix | voip-info has a very extensive list |
| 02:40.31 | Flauto | puzzled, thanks for the info, i know the very basics now, so, things are getting better |
| 02:40.48 | puzzled | Flauto: I know starting out in VoIP is a challenge. that's why it helps you if you read all you can find. the book on asteriskdocs.org (or buy a hardcopy), voip-info.org and the mailinglist archives are a wealth of info |
| 02:40.52 | klydal | you guys mostly use ata's or has anyone tried wifi phones? |
| 02:41.06 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:41.13 | Qwell | mog_home!!! |
| 02:41.15 | IronHelix | generally i use ip hardphones when i can |
| 02:41.17 | Flauto | puzzled, thanks |
| 02:41.23 | mog_home | qwell!! |
| 02:41.23 | IronHelix | atas are good if you need to use existing wiring |
| 02:41.26 | puzzled | klydal: wifi phones need better battery performance but they seem to work reasonably |
| 02:41.33 | IronHelix | or devices that need an analog line |
| 02:41.34 | ManxPower | "Is that a hardphone, or are you just happy to see me?" |
| 02:41.38 | IronHelix | hahah |
| 02:41.40 | Qwell | puzzled: from what I've seen, they have better battery than a cellphone |
| 02:41.58 | klydal | well I don't think you can use them to log into certain wifi hotspots |
| 02:42.01 | puzzled | Qwell: did you steal any of my shrooms? |
| 02:42.08 | *** join/#asterisk SwK (n=SwK@dpc6745230018.direcpc.com) |
| 02:42.13 | IronHelix | klydal- generally no, esp if the hotspot requires web login |
| 02:42.27 | IronHelix | also most wisip phones dont support WPA, only WEP |
| 02:42.28 | klydal | I could just get softphone on a pda I guess |
| 02:42.31 | *** join/#asterisk dfriend (n=dfriend@69.89.168.17) |
| 02:42.37 | IronHelix | linksys is cooking one that has WPA but no date on when it'll be out |
| 02:42.48 | IronHelix | theres a few IAX clients for PDAs, works better with NAT |
| 02:42.54 | puzzled | Qwell: seriously battery life on those babies is up for serious imprvement |
| 02:43.06 | Qwell | puzzled: sure, but let's start with cellphones |
| 02:43.09 | puzzled | IronHelix: wifi phone? |
| 02:43.25 | Qwell | get me more than 2 hours of talk time, and then we can talk about improving the 20+ hours on a wifi phone |
| 02:43.32 | klydal | I noticed most say wireless b only |
| 02:43.35 | LostFrog | They need to put nuclear cells in those. |
| 02:43.39 | IronHelix | yeah, i sometimes call them wisip phones even tho thats a pulver brand name |
| 02:43.42 | LostFrog | 100 years of power. :) |
| 02:43.43 | puzzled | Qwell: don't know about US cdma phones but my gsm phone works for many days (given my usage) |
| 02:43.43 | Qwell | klydal: well, yeah...whats the use in g? |
| 02:43.45 | IronHelix | wifi phone that uses SIP |
| 02:43.55 | Qwell | puzzled: well, sure, mine too, but if I actually use it, I get 2 hours tops |
| 02:44.00 | klydal | well I use a where I live |
| 02:44.05 | klydal | everyone has a wirless g/b |
| 02:44.07 | tzanger | RoyK: you're just making a fool of yourself now |
| 02:44.21 | RoyK | tzanger: no. i'm not...... |
| 02:44.24 | tzanger | your "am I banned" crossposts are comign through just fine |
| 02:44.27 | puzzled | Qwell: than your phone simply sucks. iirc I get at least 4 hours talk time on my gsm phone |
| 02:44.37 | RoyK | tzanger: i'm out of the -users list |
| 02:44.37 | tzanger | if you would have been banned you'd have been banned across the board, don't you think? |
| 02:44.46 | RoyK | fuck you, sir |
| 02:45.05 | tzanger | it's not every day I get referred to as 'sir' and told to fuck off :-) |
| 02:45.12 | klydal | too bad blue tooth isn't more popular in the states |
| 02:45.17 | RoyK | banning me from the users list is a bad thing |
| 02:45.25 | tzanger | I very much doubt you've been banned |
| 02:45.40 | RoyK | they have indeed stopped all my posting to that list |
| 02:45.43 | Qwell | tzanger: You weren't told to "fuck off". Just to fuck in general. |
| 02:45.54 | Qwell | or something |
| 02:45.57 | tzanger | you choose to ignore mog_home when he says that the lists have been censored in the past but only due to legal constraints on g729 binaries |
| 02:46.13 | puzzled | with the growth of the world population I'd prefer ppl to tell others to just die |
| 02:46.15 | tzanger | you also chose to ignore mog_home when he said that the one guy who is in charge of the lists is currently travelling and isn't anywhere near online |
| 02:46.34 | tzanger | RoyK: you also choose to ignore me when I say I've put many digium-critical posts to the list and not been banned |
| 02:46.45 | ManxPower | I should export my /ignore list. |
| 02:47.03 | mog_home | and there are many still in the log |
| 02:47.05 | tzanger | RoyK: Further, you ignore me when I say that you could have had your name throttled due to frequent posts, which I imagine is a measure in effect on a list as large as this one |
| 02:47.09 | mog_home | i mean google digium sucks |
| 02:47.12 | mog_home | youll find it |
| 02:47.33 | mog_home | in the mailing list |
| 02:47.43 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 02:47.45 | mog_home | digium does not enjoy being a bad guy |
| 02:47.46 | tzanger | in fact there are hundreds of anti-digium messages on -users... why would they ban you? |
| 02:47.48 | puzzled | RoyK: why don't you give it a day or two and check it again. and please stay polite. no need to be rude |
| 02:47.55 | FuriousGeorge | hey all |
| 02:47.55 | RoyK | tzanger: i've been posting regularly, quite more frequently than this |
| 02:48.06 | puzzled | hi FuriousGeorge |
| 02:48.09 | tzanger | RoyK: yes, I post quite a bit too |
| 02:48.12 | RoyK | puzzled: no need to be poite either..... |
| 02:48.20 | justinu | lol |
| 02:48.23 | puzzled | RoyK: ok try neutral then :) |
| 02:48.26 | ManxPower | mog_home, but..but..but...what about Mark's badboy image! It's all a sham??? |
| 02:48.28 | puzzled | ack |
| 02:48.28 | justinu | that's one way to look at it, i guess |
| 02:48.33 | mog_home | lol |
| 02:48.36 | tzanger | what I'm saying though is that any number of factors could be contributing to what's happening... why not take a more balanced approach instead of jumping to conclusions? |
| 02:48.40 | LostFrog | Are we still squabling about this? |
| 02:48.43 | mog_home | mark is one wild and crazy guy |
| 02:48.48 | puzzled | tzanger: agree |
| 02:48.54 | tzanger | and cute |
| 02:48.54 | tzanger | er |
| 02:48.56 | tzanger | dammit |
| 02:48.57 | harryvv | some one message me and give me a ball park figure what the entire cost of setting up a voip service woud be. Im sure the bank would want to know. |
| 02:48.57 | tzanger | smart |
| 02:49.06 | tzanger | harryvv: :-) |
| 02:49.10 | Qwell | harryvv: $18 million dollars. |
| 02:49.13 | LostFrog | wow.. that's vague, harryvv. |
| 02:49.14 | harryvv | hehe |
| 02:49.18 | puzzled | mog_home: that had his laptop stolen in Amsterdam cause someone told him it was safe to leave it in the trunk :) |
| 02:49.21 | ManxPower | harryvv, 2 million US, at least |
| 02:49.23 | supaigtr | harryv: You mean CLEC and everything? |
| 02:49.25 | *** join/#asterisk simoncion (n=simoncio@user-24-236-84-62.knology.net) |
| 02:49.25 | konfuzed | great, now I get; |
| 02:49.26 | mog_home | ouch |
| 02:49.29 | konfuzed | chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this function) |
| 02:49.29 | konfuzed | chan_zap.c: In function `load_module': |
| 02:49.29 | konfuzed | chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from incompatible pointer type |
| 02:49.29 | konfuzed | chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from incompatible pointer type |
| 02:49.29 | konfuzed | make[1]: *** [chan_zap.o] Error 1 |
| 02:49.30 | konfuzed | make[1]: Leaving directory `/usr/src/asterisk/channels' |
| 02:49.31 | IronHelix | i'll do it for 17mil |
| 02:49.32 | konfuzed | make: *** [subdirs] Error 1 |
| 02:49.37 | mog_home | that sucks |
| 02:49.38 | Qwell | ~pb |
| 02:49.39 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 02:49.41 | tzanger | konfuzed: you're still on this? |
| 02:49.44 | harryvv | sup, even just have a wholsaler take care alot of the routing i dont care |
| 02:49.52 | LostFrog | konfuzed: did you install libpri first? |
| 02:49.53 | tzanger | ditch those debian packages, and go get CVS HEAD of libpri, asterisk and zaptel |
| 02:49.53 | konfuzed | tzanger: its chokin me bad |
| 02:49.56 | puzzled | konfuzed: install libpri & libpri-devel |
| 02:50.00 | ManxPower | harryvv, Any kind of telecom these days is expensive to get into if you want to do it right. |
| 02:50.10 | harryvv | manx I know |
| 02:50.24 | Qwell | ManxPower: psh...do it right? telecom? :) |
| 02:50.27 | harryvv | im looking at smaller markets. |
| 02:50.31 | puzzled | so why isn't there an Asterisk Venture Capital Foundation?! |
| 02:50.32 | Qwell | those two are mutually exclusive |
| 02:51.01 | ManxPower | puzzled, Um, because only crazy people start a company in telecom? |
| 02:51.15 | puzzled | ManxPower: yeah like my venture with Vocalis |
| 02:51.26 | supaigtr | telecom is hard. You have to be crazy, like me, to jump in. |
| 02:51.36 | puzzled | or jerjer |
| 02:51.46 | Qwell | we're all a bit crazy...we have to be |
| 02:51.46 | harryvv | yea mabey so |
| 02:51.51 | ManxPower | There seem to be a fairly high number of smart crazy people. |
| 02:51.51 | Qwell | some just...more than others |
| 02:51.57 | harryvv | Im talking just strait voip |
| 02:53.02 | konfuzed | I did libpri already but I'll do it again and make sure that libpri devel is on the go too |
| 02:53.13 | *** join/#asterisk Kort (n=james@65.211.216.202) |
| 02:53.27 | puzzled | if I find a zillion dollars for a VoIP telco venture I will offer jobs to all * guru's with cool perks like stroopwafels and shrooms |
| 02:53.34 | Kort | hi, I had some questions about getting callerID working |
| 02:53.36 | tzanger | haha |
| 02:54.15 | IronHelix | puzzled- sign me up for that, although can i trade my shrooms for a bunch of free ip phones? |
| 02:54.16 | LostFrog | What is a stroopwafel? |
| 02:54.22 | puzzled | Kort: the good side of it not working is that your soon to be ex has no clue it's you that's calling :) |
| 02:54.22 | supaigtr | Important things in VOIP is a locked market (small) and having control over last mile to those users. |
| 02:54.29 | mog_home | so whats up tonight people |
| 02:54.42 | asterboy | my sleave |
| 02:54.43 | Kort | puzzled: haha, unfortunately this is for a business.. |
| 02:54.58 | puzzled | LostFrog: dutch waffel with caramel in it that's actually legal in other countries too |
| 02:54.58 | LostFrog | My boss's blood pressure, mog_home. |
| 02:55.06 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 02:55.10 | mog_home | whats wrong lostfrog |
| 02:55.17 | justinu | puzzled: yummy |
| 02:55.20 | LostFrog | Just deployed the PBX today. |
| 02:55.27 | mog_home | uh o |
| 02:55.29 | LostFrog | Lots of glitches to work through. |
| 02:55.31 | mog_home | whats wrong with it |
| 02:55.34 | LostFrog | Most snom stuff. |
| 02:55.42 | LostFrog | People getting hung up on. |
| 02:55.43 | Kort | so, I've got callerid=asreceived in my zapata.conf |
| 02:55.45 | puzzled | LostFrog: weren'y you supposed to test it before deploying it? |
| 02:55.45 | IronHelix | voicemail problems? |
| 02:55.47 | justinu | uh-oh |
| 02:55.52 | Kort | but the ${CALLERID} variable is always blank |
| 02:55.52 | Qwell | puzzled: never |
| 02:55.52 | justinu | dropped calls? |
| 02:55.59 | puzzled | Qwell: :) |
| 02:56.02 | LostFrog | puzzled: I did.. But I had no problems using it. |
| 02:56.08 | Kort | (the calls are coming in over a Sangoma T1 card) |
| 02:56.16 | LostFrog | it's only in harried volume that problems came up. |
| 02:56.23 | tengulre | I have a problem! |
| 02:56.25 | LostFrog | I think they are all fixed now. |
| 02:56.33 | mog_home | yay |
| 02:56.45 | puzzled | Kort: turn on debug and study that if the telco provides it |
| 02:56.50 | LostFrog | transfer_on_hook is a bad thing on snoms. |
| 02:57.29 | puzzled | LostFrog: does it differ from transferring with the handset lifted? |
| 02:57.46 | justinu | lostfrog: why did you guys choose snom? (curious) |
| 02:57.49 | tengulre | does the Diguim's cards support china telecom? |
| 02:57.49 | mog_home | man it was such a busy today, i havent even read slashdot |
| 02:57.58 | Qwell | heathen! |
| 02:58.03 | puzzled | hahaha |
| 02:58.06 | Kort | puzzled: how do I turn on real debugging in asterisk? anything better than -vvvvvv and write "debug" to log file? |
| 02:58.15 | Qwell | /. is priority n-1 |
| 02:58.18 | mog_home | whats that tengulre |
| 02:58.22 | puzzled | Kort: set debug 10 or something like that |
| 02:58.22 | Flauto | puzzled, do you know vbuzzer? i am trying so hard to make it to work my asterisk |
| 02:58.25 | Flauto | but it does not |
| 02:58.32 | LostFrog | puzzled: if you have a call on hold and hang up on a call, it transfers one call to the other. :) |
| 02:58.35 | puzzled | Flauto: never heard of it. what is it? |
| 02:58.56 | LostFrog | snom: I like the feel and the working call presence. |
| 02:59.12 | LostFrog | justinu, even. |
| 02:59.15 | puzzled | LostFrog: don't the cisco's and polycom's have that too? |
| 02:59.17 | Flauto | puzzled, it is a service based in toronto, canada, it provides free did and free calling within 416 area code |
| 02:59.20 | Kort | puzzled: should the debug output be in the CLI interface? |
| 02:59.24 | tengulre | mog_home: I m in china, I want to buy a diguim card, but I don't know it's support our telecom standrad? |
| 02:59.25 | Kort | there doesn't seem to be much there.. |
| 02:59.26 | puzzled | Kort: yes |
| 02:59.34 | IronHelix | kort try something like pri debug |
| 02:59.35 | Flauto | i have a few friends in toronto so i signed up |
| 02:59.35 | mog_home | probably will |
| 02:59.43 | mog_home | it can set diff impeds. |
| 02:59.44 | mog_home | etc |
| 02:59.54 | Flauto | it provides a soft phone for its service |
| 03:00.01 | mog_home | but the disconnect i dont know |
| 03:00.06 | puzzled | Flauto: no idea. check voip-info.org or google. |
| 03:00.12 | Flauto | i did |
| 03:00.14 | mog_home | i have no word of people using it in china |
| 03:00.16 | Flauto | and found some info |
| 03:00.17 | konfuzed | well it seems to be processing further so maybe it will finish without an error this time |
| 03:00.18 | mog_home | i imagine people are |
| 03:00.37 | Flauto | but it is showing the service is registered but i can not recieve any call |
| 03:00.39 | tengulre | mog_home: :( |
| 03:00.43 | Kort | I don't think I'm using PRI |
| 03:00.49 | Flauto | and no voice trafic when i call out |
| 03:00.54 | mog_home | but you might ask on mailing list |
| 03:00.55 | puzzled | Flauto: maybe that's your firewall not letting it in |
| 03:00.58 | mog_home | i dont always know it all |
| 03:00.59 | justinu | lostfrog: so how many snoms? |
| 03:01.11 | Flauto | even i put the computer in the dmz, it i sstill not working |
| 03:01.14 | tengulre | yes! I see! |
| 03:01.18 | LostFrog | currently 11. |
| 03:01.34 | justinu | ok, i have about 15 polycoms |
| 03:01.34 | Qwell | so, what do you guys do about distributed *> |
| 03:01.45 | Qwell | ie; in an env where 1 box just won't cut it |
| 03:01.49 | justinu | i use ser to route inbound calls |
| 03:01.50 | LostFrog | polycoms are too expensive for us. |
| 03:01.52 | justinu | and outbound |
| 03:02.04 | Qwell | justinu: that to me? |
| 03:02.05 | puzzled | Qwell: get another box and use DNS SRV stuff and/or a smart dialplan |
| 03:02.06 | mog_home | i want to try that new linksys phone |
| 03:02.12 | justinu | Qwell: sorta |
| 03:02.18 | mog_home | that looks like a cisco 7940 |
| 03:02.18 | Qwell | puzzled: can't do anything dns |
| 03:02.21 | mog_home | it looked hot |
| 03:02.36 | Qwell | puzzled: but what about things like queues? |
| 03:02.53 | hhoffman | has anyone used FXS->FXO converters? |
| 03:03.01 | Qwell | converters? how silly |
| 03:03.11 | puzzled | Qwell: put a SER server in front of it with no moving parts and a dual power supply. will work forever. no idea bout queues |
| 03:03.14 | mog_home | you mean like a tdm400p.... |
| 03:03.24 | Qwell | mog_home: gotta start pimping the 2400 |
| 03:03.27 | Flauto | puzzled, any idea about it? |
| 03:03.34 | puzzled | Flauto: nope |
| 03:03.43 | mog_home | lol |
| 03:03.49 | mog_home | i do love it |
| 03:03.51 | Flauto | thanks puzzled, |
| 03:05.05 | Qwell | puzzled: What if you aren't using SIP? |
| 03:05.10 | LostFrog | One thing about the snoms, I wish there was a way to push settings to it. |
| 03:05.13 | ManxPower | Qwell, I've been told the 2400 is a totally different design from previous digium cards. |
| 03:05.29 | Qwell | ManxPower: probably. |
| 03:05.31 | LostFrog | like http://<ip>/settings.php?source=http://server/snom320.cfg |
| 03:05.36 | drumkilla | ManxPower: indeed, it is |
| 03:05.37 | ManxPower | But we are comitted to T-1s now. |
| 03:05.42 | Kort | puzzled: any other ideas for debugging? |
| 03:06.27 | ManxPower | We had another instance of all FXS ports on our 3 TDM400Ps in one of our Asterisk systems not working again today. |
| 03:06.39 | ManxPower | Seems to happen once a month or so |
| 03:06.55 | docelm0 | Anybody in here do A-Z termination? |
| 03:06.57 | puzzled | beats the 21 days or so thing earlier |
| 03:07.22 | puzzled | docelm0: voip-info.org has a long list |
| 03:07.34 | asterboy | ManxPower, want to sell your FXS ports? Get new ones and I'll buy your defective one. |
| 03:07.59 | ManxPower | asterboy, You'd have to take all three. |
| 03:07.59 | docelm0 | eh.. figured I would ask |
| 03:08.06 | Kort | argh, I can't seem to get any low-level debug info |
| 03:08.44 | asterboy | Depends on the price. |
| 03:09.10 | justinu | qwell, there's no way to remotely provision them? |
| 03:09.13 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 03:09.14 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 03:09.24 | justinu | s/qwell/lostfrog |
| 03:10.01 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 03:10.14 | LostFrog | There is.. but I don't want the phones in India provisioned in the United States, and I don't have a decent server in India. |
| 03:10.35 | Kort | anyone tried to debug a sangoma T1 card? |
| 03:10.49 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 03:11.43 | LostFrog | I just want a one-time shot thing. |
| 03:11.58 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
| 03:12.34 | justinu | i'm kinda surprised you can't upload a settings file via the web interface |
| 03:12.39 | Kort | I can't get callerid off of this damn thing |
| 03:12.49 | Kort | and I haven't found a way (yet) to get any meaningful debug info. |
| 03:13.49 | asterboy | 300 users in here on average...thats darn good. |
| 03:14.06 | LostFrog | Yeah.. but there is a 100/1 lurk rate. :) |
| 03:14.16 | mog_home | lol |
| 03:14.16 | *** part/#asterisk damned (n=vpol@prior.lanck.net) |
| 03:16.13 | wasim | Kort: pri debug span 1 |
| 03:16.51 | infinity1 | is voipjet down? |
| 03:17.36 | konfuzed | so now it crapped out on chan_modem so I deleted /etc/asterisk/modem.conf and make clean ; make ; make install |
| 03:17.37 | Kort | I'm not using pri |
| 03:17.41 | Math[laptop] | infinity1, no |
| 03:17.43 | Kort | (or it says there's no PRI on span 1) |
| 03:17.52 | Kort | I have a voice t1. |
| 03:17.57 | Kort | through a sangoma card. |
| 03:19.11 | Kort | so I'm not sure how i can debug this at the moment |
| 03:20.06 | infinity1 | Math[laptop]: it was working for me this morning, and now its not. argh |
| 03:20.24 | Math[laptop] | just made a call thru it... which server are you on? |
| 03:20.26 | Math[laptop] | east-coast? |
| 03:20.52 | infinity1 | Math[laptop]: i just tried 64.34.45.100 |
| 03:20.59 | *** join/#asterisk bweschke (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 03:21.21 | wunderkin | hey bj |
| 03:21.22 | *** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
| 03:21.22 | Math[laptop] | infinity1, Im on 216.118.117.46 and it worked fine |
| 03:21.32 | bweschke | hey wunderkin, what's up? :) |
| 03:21.38 | wunderkin | nothin |
| 03:22.06 | wunderkin | hows lv and ip4it |
| 03:22.22 | bweschke | not bad.. sitting in the developer's fishbowl now in the booth |
| 03:22.39 | asterboy | watch a-z termination services rocket when WiMAX gets mainstream. |
| 03:22.42 | infinity1 | hm. mine accepts the call and hangs up |
| 03:22.43 | infinity1 | <PROTECTED> |
| 03:22.43 | infinity1 | <PROTECTED> |
| 03:22.43 | infinity1 | <PROTECTED> |
| 03:22.47 | infinity1 | i'll try yours |
| 03:23.37 | infinity1 | strange. that didn't work either |
| 03:23.39 | Kort | so what's the best way to debug a voice t1 that doesn't use PRI? |
| 03:23.49 | tzanger | Kort: what's it (not) doing? |
| 03:23.57 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 03:24.06 | Kort | tzanger: ${CALLERID} in asterisk is always blank on incoming calls |
| 03:24.15 | tzanger | Kort: how is your provider sending CID |
| 03:24.27 | Kort | tzanger: I'm not sure - I'd have to check with MCI |
| 03:24.33 | tzanger | well that's step #1 |
| 03:25.02 | Kort | and then after I find that out? |
| 03:25.10 | tzanger | depends on what they say, of course. |
| 03:25.34 | *** join/#asterisk alvariux (n=unky@201.155.166.186) |
| 03:25.39 | alvariux | hello |
| 03:25.52 | asterboy | hola |
| 03:25.55 | alvariux | somebody have build iaxclient? |
| 03:26.03 | alvariux | im having some errors |
| 03:27.22 | alvariux | im trying to build testcall but im getting testcall-jb.c:51: error: el tipo matriz tiene tipo de elemento incompleto |
| 03:27.22 | alvariux | testcall-jb.c: En la función ?jm_init?: |
| 03:27.22 | alvariux | testcall-jb.c:109: aviso: el puntero que apunta en el paso del argumento 3 de ?getsockname? difiere en signo |
| 03:27.28 | Kort | tzanger: yeah at this point I'm wondering if they even send it over the T1. |
| 03:27.49 | alvariux | sorry my system is in spanish |
| 03:28.03 | justinu | kort: that's not a pri? |
| 03:28.18 | Kort | justinu: nope, PRI is not running |
| 03:28.32 | justinu | what kind of signalling? e&m wink start? |
| 03:28.38 | alvariux | asterboy hablas ingles |
| 03:28.40 | infinity1 | http://pastebin.ca/28743 |
| 03:28.45 | alvariux | español |
| 03:28.48 | infinity1 | Math[laptop]: can you check my iax debug? |
| 03:28.55 | Kort | yeah em_w |
| 03:28.56 | Math[laptop] | sure |
| 03:29.06 | infinity1 | it seems voipjet hangs up on me after it accepts the call. |
| 03:29.23 | justinu | kort: they usually don't have callerid on an e&m trunk |
| 03:29.29 | justinu | you gotta ask special for it |
| 03:29.32 | Kort | awesome. |
| 03:29.36 | Kort | damn them. |
| 03:29.38 | justinu | kort: tell them you want "feature group d" |
| 03:29.57 | justinu | do you have MF tone receivers? |
| 03:30.02 | Kort | yeah this is what I get for taking over crap that someone else started.. |
| 03:30.14 | Kort | I don't. |
| 03:30.27 | justinu | ok, then fgd won't work, since it's MF not DTMF |
| 03:30.36 | Kort | well hmm lemme think |
| 03:30.36 | infinity1 | the big line i see is: Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP |
| 03:30.46 | asterboy | un poco |
| 03:31.11 | Kort | let me check |
| 03:32.17 | Math[laptop] | infinity1, does the target # ring? |
| 03:32.37 | infinity1 | Math[laptop]: no. |
| 03:33.07 | Kort | justinu: would my Sangoma A101 card have em? |
| 03:33.15 | infinity1 | the call is accepted at Timestamp: 00081ms |
| 03:33.21 | infinity1 | and hangs up at Timestamp: 00117ms |
| 03:33.36 | infinity1 | i don't think you'll fit a ring in 36 ms :) |
| 03:33.54 | asterboy | como se lama usted? |
| 03:34.06 | Math[laptop] | lol |
| 03:34.13 | justinu | kort: not sure about that kind of hardware |
| 03:34.53 | infinity1 | my account has $.71 in it. |
| 03:35.01 | konfuzed | hm deleting /var/lib/asterisk/modules/chan_ modem.so let me get past this error and next is chan_capi.so but I doubt that should be deleted so easily as chan_modem.so |
| 03:35.23 | Qwell | infinity1: I've got you beat. -$5.47 |
| 03:35.25 | wunderkin | infinity1, i think your account has to have more than that.. unless they make an exception for your account |
| 03:35.34 | Qwell | oh, voip account... ;/ |
| 03:35.35 | justinu | wow, you guys are poor |
| 03:35.39 | konfuzed | will asterisk run without chan_capi.so |
| 03:35.54 | asterboy | government thinks I owe them 150,000 |
| 03:35.58 | asterboy | thats poor |
| 03:36.01 | sbingner | konfuzed: sure, as long as you're not using capi channels |
| 03:36.06 | mog_home | ues konfuzed |
| 03:36.11 | Math[laptop] | infinity1, you need to have funds for 2 hours of talking |
| 03:36.37 | infinity1 | Math[laptop]: for realz? |
| 03:36.38 | konfuzed | what uses capi channels ( my lingo is on holiday ) |
| 03:36.46 | Math[laptop] | which is, for us&canada, 120*0.013 = 1.56$ |
| 03:36.48 | Qwell | konfuzed: capi channels use capi channels |
| 03:36.57 | infinity1 | Math[laptop]: shit. it helps if they would tell me that. |
| 03:37.00 | Math[laptop] | infinity1, yeah, and that for each concurrent call you want to make |
| 03:37.04 | Math[laptop] | infinity1, they tell you that |
| 03:37.18 | LostFrog | konfuzed: ISDN |
| 03:37.19 | infinity1 | Math[laptop]: where? |
| 03:37.33 | Math[laptop] | "Here's how it works: Say you have a balance of ten (10) dollars. When your first call comes in, two hours worth of credit is temporarily frozen. So, if you are calling a destination that costs 2 cents a minute, that means $2.40 (2 cents/min x 120 minutes) has been frozen in your account. " |
| 03:37.34 | infinity1 | Math[laptop]: its okay. you did. i looked on their website. don't see it. |
| 03:37.37 | konfuzed | Qwell: how qould you describe that with out using the acronym capi |
| 03:37.38 | Math[laptop] | in the FAQ, last item, for concurrent calls |
| 03:37.41 | konfuzed | isdn is |
| 03:37.43 | konfuzed | good |
| 03:37.45 | Qwell | ~capi |
| 03:37.46 | jbot | methinks capi is Common ISDN Application Programming Interface. See http://www.capi.org for more info. |
| 03:37.48 | konfuzed | as in the x100p |
| 03:37.59 | mog_home | ~qwell |
| 03:38.01 | jbot | rumour has it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 03:38.05 | Qwell | :D |
| 03:38.06 | mog_home | it should same master of the jbot |
| 03:38.17 | mog_home | err say |
| 03:38.18 | LostFrog | no.. x100p is PSTN. |
| 03:38.19 | Math[laptop] | infinity1, voxee is cheaper for us&canada tho |
| 03:38.20 | LostFrog | analog |
| 03:38.24 | infinity1 | Math[laptop]: you would think it they would add some php code: if ($amount < 2.40) { echo "you can't make a call"; } |
| 03:38.36 | LostFrog | x100p uses chan_zap |
| 03:38.39 | konfuzed | so capi actually requires a physical isdn modem then |
| 03:38.49 | infinity1 | Math[laptop]: thanks. i'll go their. you would think voip and teliax could improve their interface with about 2 mins of coding. heh |
| 03:38.50 | Math[laptop] | infinity1, they could add a "low-fund warning " |
| 03:39.13 | konfuzed | is it required by some weird setup not using commercial isdn services |
| 03:39.36 | Math[laptop] | infinity1, well it depends where you wanna call, sometimes voipjet is more advantageous than voxee, and vice-versa, you need smth capable of doing LCR |
| 03:39.53 | asterboy | nice list of providers: http://www.iptelephony.org/GIP/providers/ |
| 03:40.24 | infinity1 | Math[laptop]: smth? |
| 03:40.27 | Math[laptop] | something |
| 03:40.35 | infinity1 | heh. ;) |
| 03:40.41 | konfuzed | some how I got the impression that capi was essentiall to core functioning even with out paying for services called ISDN from telco |
| 03:41.29 | Math[laptop] | why? you can use Zap |
| 03:41.41 | konfuzed | so there are no issues to arise from deleteing the capi modules then |
| 03:41.49 | infinity1 | works now! yay. |
| 03:41.52 | konfuzed | maybe it was set up for fax some how |
| 03:41.58 | infinity1 | thanks for helping me RTFM :) |
| 03:42.17 | Math[laptop] | infinity1, np :) |
| 03:43.18 | tainted_ | how do i connect two live sip channels together w/o using meetme? |
| 03:43.53 | mog_home | transfer |
| 03:43.54 | Math[laptop] | tainted_, call parking? |
| 03:44.12 | konfuzed | ok all seven of them are now gone |
| 03:44.21 | Math[laptop] | mog_home, you can transfer 2 incoming calls together? |
| 03:44.35 | mog_home | with manager i think |
| 03:44.55 | tainted_ | Math[laptop] have you done it successfully with call parking? |
| 03:44.57 | Math[laptop] | uhm the only manager I used was an old win32 port of gastman |
| 03:45.09 | Math[laptop] | tainted_, nope, but its doable |
| 03:46.26 | konfuzed | ok now Im up to pbx_gtkconsole |
| 03:46.39 | konfuzed | how can i udpate pbx_gtkconsole |
| 03:46.59 | justinu | off topic: anyone ever use rrdtool? |
| 03:47.53 | konfuzed | when I finish this box I get to another one |
| 03:48.06 | konfuzed | should be less of a pain next time around |
| 03:48.34 | konfuzed | ive dont it four times before actually but this one is a tad messed |
| 03:48.58 | konfuzed | im gittin there |
| 03:49.07 | konfuzed | wait till ive done a dozen |
| 03:49.21 | konfuzed | how can i udpate pbx_gtkconsole |
| 03:51.30 | konfuzed | hey there is no /etc/asterisk/modules.conf file |
| 03:51.43 | konfuzed | isnt that suppose to be created with the make install |
| 03:51.57 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 03:52.15 | Math` | no |
| 03:52.25 | Math` | but "make samples" installs it.. but beware this is going to erase your actual configuration |
| 03:52.35 | Math` | so you better copy it off the source tree |
| 03:55.17 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 03:55.46 | kuku5 | anyone using an operator panel for asterisk on more than 40 extensions ? |
| 03:57.45 | LostFrog | Ok.. stupid question.. I know there are no US area codes starting with 1, but is one starting with 11 legal? |
| 03:57.50 | asterboy | anyone use this one: http://www.talkvoip.ca/phoneservice.html |
| 03:58.05 | LostFrog | duh.. forget I asked that. |
| 03:58.23 | LostFrog | Please. |
| 03:58.27 | Math` | LostFrog: hehehe |
| 03:58.32 | LostFrog | Brain fart. |
| 03:58.36 | Math` | yeah |
| 03:58.47 | Math` | you can check the nanpa for area code stuff tho :P |
| 03:58.51 | LostFrog | I know. |
| 03:58.57 | LostFrog | I was looking at it. |
| 03:59.12 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 04:00.45 | konfuzed | well i kicked it enough and it runs now |
| 04:01.21 | Kort | the fact that my phones don't connect immediately after dialing an internal 4-digit extension means that I have a pattern match that is still outstanding right? |
| 04:02.12 | rajiv | i have to dial NNNN# to get my phones to connect |
| 04:02.22 | Kort | yeah so do I |
| 04:02.23 | Kort | and I hate it. |
| 04:02.27 | LostFrog | I feel scared putting 911 in my dialplan.. |
| 04:02.33 | LostFrog | How do i test it? :( |
| 04:02.33 | rajiv | i dont mind it but my wife can't stand it |
| 04:02.40 | konfuzed | but I call the number that the x100p is plugged into and * does not seem to notice |
| 04:02.43 | rajiv | LostFrog: test it with 411 |
| 04:02.44 | Kort | I'm trying to fix that issue.. |
| 04:02.55 | Math` | LostFrog: I tested 911, and confirmed my address on file with an operator |
| 04:02.58 | LostFrog | rajiv: what phones do you have? |
| 04:03.09 | rajiv | innomedia mta 3308 |
| 04:03.14 | hhoffman | anyone help with this error? NOTICE[12632]: pbx.c:1716 pbx_extension_helper: Cannot find extension context 'default' |
| 04:03.19 | hhoffman | why do I need a default context if I have the SIP number in another context? |
| 04:03.20 | LostFrog | Hmm.. can't help you there. |
| 04:03.23 | rajiv | apparently no one else has these phones |
| 04:03.47 | rajiv | the phones have a digit match thing but i haven't set them up |
| 04:03.51 | LostFrog | Thanks for the 411 sugestion. |
| 04:04.39 | LostFrog | All working. |
| 04:05.42 | LostFrog | I just wanted to make sure 911 was going out the landlines. |
| 04:05.42 | Kort | unfortunately I don't see any pattern that would prevent it from matching my extensions immediately. |
| 04:07.11 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:07.11 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:08.37 | docelm0 | Termination .009 something termination no commits or minimums.. FREE .25c to try.. http://www.plainvoip.com |
| 04:09.14 | LostFrog | "Football City?" |
| 04:09.30 | deezed | docelm0 you run it? |
| 04:09.43 | docelm0 | more or less |
| 04:10.02 | Math` | docelm0: is there a price list somewhere? |
| 04:10.04 | LostFrog | Only NYC DIDs? |
| 04:10.13 | docelm0 | Much more coming.. Just ordered.. |
| 04:10.13 | Kort | is there any way to see what dialplan rules asterisk is trying to match? |
| 04:10.30 | LostFrog | docelm0: any chance of DC area DIDs? |
| 04:10.38 | docelm0 | Yes |
| 04:10.44 | docelm0 | I just ordered 200 from around the country |
| 04:10.52 | docelm0 | I told them gimme a few from all areas |
| 04:12.30 | *** join/#asterisk freespace-in (n=special@ppp-70-225-137-82.dsl.ipltin.ameritech.net) |
| 04:12.34 | asterboy | I'd like to setup a company like your in Edmonton. |
| 04:12.50 | freespace-in | anyone using a cisco router with mgcp? |
| 04:12.57 | docelm0 | Why? |
| 04:12.59 | docelm0 | SIP |
| 04:13.01 | freespace-in | i'm trying to use the fxo ports and can't seem to dialout |
| 04:13.21 | morale | fxo ports to dialout with a analog phone? |
| 04:13.23 | docelm0 | What IOS and Model are you using? |
| 04:13.35 | freespace-in | mc3810 ios 12.2.15T17 |
| 04:13.38 | freespace-in | w/voice |
| 04:13.45 | konfuzed | ah well ive had enough of asterisk for now |
| 04:13.51 | freespace-in | fxo to dialout to the phone company |
| 04:13.58 | konfuzed | im gonna install that edubuntu amd 64 |
| 04:14.01 | *** join/#asterisk jjones (n=jjones@adsl-223-72-14.aep.bellsouth.net) |
| 04:14.07 | konfuzed | and then try asterisk on that |
| 04:14.19 | konfuzed | we'll see how it goes |
| 04:16.21 | freespace-in | any usable sip configs for a cisco? |
| 04:16.57 | docelm0 | I could do it in my sleep |
| 04:17.00 | docelm0 | What do you wanna know? |
| 04:17.32 | freespace-in | i have a fxs and and fxo port i want to use |
| 04:17.40 | freespace-in | i want to send _NXXXXXX out fxo |
| 04:17.46 | freespace-in | i have extension 2010 on fxs |
| 04:18.09 | freespace-in | sip is new to me, i'm more comfortable in CCM world with MGCP and SCCP |
| 04:18.22 | freespace-in | so Sip is a whole different beast |
| 04:19.16 | docelm0 | SIP is cake to setup on a cisco.. Just like any other.. As long as you can configure a dialplan |
| 04:19.20 | docelm0 | err dialpeer |
| 04:20.14 | freespace-in | okay, great. how do i do it then? |
| 04:20.20 | freespace-in | basics |
| 04:20.23 | docelm0 | Your new to cisco right? |
| 04:20.36 | freespace-in | nope |
| 04:20.49 | freespace-in | but i'm more R&S |
| 04:20.56 | docelm0 | Then configure your dialpeer.. Its just the same as H323 or anything else |
| 04:20.56 | freespace-in | i'm new to asterisk |
| 04:21.00 | stbain | freespace-in: read this: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 04:21.40 | docelm0 | stbain he is using a cisco box not phone |
| 04:21.43 | stbain | ahhhh |
| 04:22.47 | freespace-in | yeah, a router. |
| 04:22.51 | freespace-in | with voice ports |
| 04:22.54 | *** part/#asterisk alvariux (n=unky@201.155.166.186) |
| 04:23.25 | freespace-in | docelm0, you've gotta have a running config that can be grepped easily |
| 04:23.33 | freespace-in | i haven't seen much out there |
| 04:23.35 | docelm0 | Not for your machine |
| 04:23.43 | freespace-in | why? |
| 04:23.46 | docelm0 | simpler just to config it.. |
| 04:23.48 | freespace-in | cuz its mc3810? |
| 04:23.56 | freespace-in | makes no difference |
| 04:24.02 | docelm0 | I use Cisco 3600, 5350, 5400, 5850 series |
| 04:24.04 | freespace-in | its like 26xx |
| 04:24.06 | docelm0 | uhh HELL ya it does |
| 04:24.14 | freespace-in | its still ios |
| 04:24.19 | freespace-in | just older... |
| 04:24.29 | freespace-in | the basics are still the same |
| 04:24.33 | docelm0 | exactly |
| 04:24.54 | docelm0 | but the configs are different expecially since your using POTS and not PRI |
| 04:25.14 | docelm0 | dial-peer voice 12345 pots |
| 04:25.21 | docelm0 | incoming called . |
| 04:25.47 | *** join/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca) |
| 04:25.51 | docelm0 | port xx:D |
| 04:26.07 | docelm0 | destination-pattern REGEX |
| 04:26.11 | docelm0 | there pots dialpeer |
| 04:26.12 | freespace-in | obviosly its going to have to plar since i get no digits |
| 04:26.13 | freespace-in | voice-port 1/4 |
| 04:26.13 | freespace-in | <PROTECTED> |
| 04:26.13 | freespace-in | <PROTECTED> |
| 04:26.29 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-82-80-162-23.red.bezeqint.net) |
| 04:26.39 | PoWeRKiLL | Hello |
| 04:26.42 | freespace-in | but do i have to do anything exciting under sip-ua? |
| 04:26.45 | docelm0 | I would have to see the config |
| 04:26.48 | docelm0 | no |
| 04:26.54 | docelm0 | well normally no |
| 04:26.59 | PoWeRKiLL | someone know about this error WARNING[9181]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler |
| 04:26.59 | PoWeRKiLL | Nov 15 06:25:36 WARNING[9181]: loader.c:554 load_modules: Loading module app_rxfax.so failed! |
| 04:27.03 | docelm0 | I woul have to check my 1760 |
| 04:27.14 | docelm0 | haha! |
| 04:27.20 | docelm0 | I got that error forever |
| 04:27.27 | docelm0 | did you ever use 0.0.3? |
| 04:27.37 | docelm0 | of spansp? |
| 04:27.39 | PoWeRKiLL | docelm0 no never |
| 04:28.02 | PoWeRKiLL | I'm upgrading my asterisk from 1.0.9 to 1.2 |
| 04:28.04 | docelm0 | did you copy everything in the patch in the Makefile? |
| 04:28.16 | PoWeRKiLL | Yes |
| 04:28.31 | docelm0 | including app_rxfax.so: part? |
| 04:28.37 | docelm0 | where the gcc command is? |
| 04:28.38 | Kort | any reason why if in my [incoming] dialplan I only have exten => 5478,1,Dial(${MYSIPADDR},20) that it still waits for a timeout or '#' when dialing 5478?? |
| 04:29.03 | docelm0 | It should be Dial(SIP/#{EXTEN}) |
| 04:29.07 | Kort | well yeah |
| 04:29.28 | docelm0 | Try adding _5478,1, |
| 04:29.30 | Kort | I just through shit in there for the sake of example.. |
| 04:30.29 | PoWeRKiLL | docelm0 checking |
| 04:30.38 | docelm0 | Hay frog where you @ Astri2005? |
| 04:30.45 | Kort | _5478,1, makes no different |
| 04:30.49 | Kort | difference* |
| 04:30.59 | docelm0 | Do you get an error? |
| 04:31.06 | Kort | nope, still dials fine |
| 04:31.18 | docelm0 | What kinda phone do you use? |
| 04:31.24 | Kort | cisco 7960 |
| 04:31.39 | docelm0 | Cuase there is a timeout when waiting for digits |
| 04:31.51 | docelm0 | Its in the phone not Asterisk |
| 04:31.52 | Kort | other extensions work fine (like *86XNNN) for example |
| 04:31.55 | PoWeRKiLL | docelm0 thanks it was the problem the patch didn't apply correctly |
| 04:31.58 | docelm0 | All hardphones are like that |
| 04:32.10 | kuku5 | anyone used door phones with asterisk ? |
| 04:32.29 | rajiv | Kort: maybe it is the phone? |
| 04:32.35 | Qwell | Kort: look at the phones dialplan |
| 04:33.02 | BleedingMe | If I wipe out 1.0.9 and install 1.2 beta.... should all my configs work? |
| 04:33.07 | Qwell | BleedingMe: no |
| 04:33.18 | BleedingMe | k |
| 04:33.23 | Qwell | most will, but some things simply won't work |
| 04:33.28 | docelm0 | ohh well Im off to bed.. |
| 04:33.30 | Qwell | There is a document in the tree that covers this. |
| 04:33.32 | rajiv | there is an UPGRADING doc |
| 04:33.33 | *** part/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca) |
| 04:33.35 | Kort | yeah once I find the damn thing |
| 04:33.45 | BleedingMe | cool.. i'll check it out.. thanks! |
| 04:33.47 | Qwell | Kort: xml file |
| 04:33.58 | Qwell | dialplan.xml in tftp |
| 04:34.21 | Kort | yeah see that's tricky since I didn't set up the phones |
| 04:34.28 | Kort | (btw, no one use fonality, ever) |
| 04:34.36 | tainted_ | how do i set callerid within an agi script? |
| 04:34.37 | konfuzed | well edubuntu is going smooth so far |
| 04:34.37 | Kort | (I didn't make that choice either) |
| 04:34.39 | konfuzed | ;^) |
| 04:34.41 | Kort | fucking idiots. |
| 04:35.00 | tainted_ | given my AGI object is $objAGI and the callerid i want is 1234567890 |
| 04:35.32 | BleedingMe | and would I be correct if I were under the impression that the stable 1.0.9 is going to have echo issues using the digium quad pri card? like, no matter what? seems like there's major echo problems across the board... and i guess there's some new echo cancellers in 1.2... does that sound right? |
| 04:35.40 | Kort | quit |
| 04:37.33 | morale | s |
| 04:40.29 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 04:43.27 | konfuzed | well so far so good |
| 04:43.46 | konfuzed | any *buntu seems to install easy |
| 04:44.39 | konfuzed | ive had 40 year old ladies install it on their own box with out me there |
| 04:44.56 | konfuzed | pros and cons to everything |
| 04:46.27 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:48.03 | *** join/#asterisk freespace-in (n=special@ppp-70-225-137-82.dsl.ipltin.ameritech.net) |
| 04:48.36 | Math` | how is amp? |
| 04:49.36 | konfuzed | ok system reboot |
| 04:49.48 | SkramX | icky, amp |
| 04:49.49 | SkramX | lol |
| 04:49.52 | *** join/#asterisk redax (n=redax@r6.hu) |
| 04:49.56 | redax | hi, |
| 04:50.20 | Math` | just looking for smth for users to be able to see their CDRs and at what rate it was billed |
| 04:50.25 | Math` | config is done manually |
| 04:51.48 | redax | is there a way to see the number of the incoming call somehow? |
| 04:52.29 | *** join/#asterisk graphyx_home (n=mike@c-67-169-246-4.hsd1.ut.comcast.net) |
| 04:55.30 | hypa7ia | anyone know if anyone other than Bell does single number reach in canada? |
| 04:56.53 | graphyx_home | Does express talk for windows work properly with asterisk 1.2 rc2? |
| 04:57.02 | graphyx_home | or is it just me who doesn't know what I am doing? |
| 04:59.56 | justinu | onset telecom canada seems to offer it |
| 05:00.33 | justinu | oh, guess that's just a reseller |
| 05:03.15 | konfuzed | i wonder if it'll burn dvds right off the primary install |
| 05:03.19 | konfuzed | thatd be nice |
| 05:05.14 | hypa7ia | justinu: yeah, i'm mostly seeing bell resellers |
| 05:07.52 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 05:09.16 | rajiv | i have an analog phone connected into a tdm400p. all i hear on the line is static |
| 05:09.21 | rajiv | it was working earlier today |
| 05:10.08 | rajiv | what could cause this ? |
| 05:10.10 | justinu | hypa7ia: it's also called "one number" server sometimes |
| 05:10.20 | justinu | or "find me follow me" |
| 05:10.36 | justinu | that's service, not server |
| 05:11.15 | hypa7ia | i'm being lazy... i should just get it working on my asterisk box |
| 05:11.29 | justinu | lol, in that case.... hit the books |
| 05:11.30 | Dr_Ray | anyone watching eagles cowboys? |
| 05:11.38 | rajiv | when i call zap/2 i hear it ringing on the calling phone but the analog phone does not actually ring |
| 05:11.46 | justinu | why give anymore money to a ILEC? |
| 05:12.51 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 05:17.05 | *** join/#asterisk Cin (i=cin@82-33-137-16.cable.ubr02.wiga.blueyonder.co.uk) |
| 05:17.10 | hypa7ia | justinu: indeed. especially as i've given notice of my departure from that CLEC already, so no more employee discount :p |
| 05:17.17 | rajiv | this static sounds real bad |
| 05:17.25 | Qwell | hypa7ia: you're quitting? |
| 05:17.36 | *** part/#asterisk Cin (i=cin@82-33-137-16.cable.ubr02.wiga.blueyonder.co.uk) |
| 05:17.39 | hypa7ia | Qwell: already did :) |
| 05:17.42 | Qwell | ahh |
| 05:17.48 | hypa7ia | last day's dec 2nd |
| 05:18.04 | hypa7ia | they are all sad and stuff, but oh well :) |
| 05:18.13 | Qwell | screw em |
| 05:18.19 | hypa7ia | precisely. |
| 05:18.33 | Qwell | so, didn't like it there anymore, or what? |
| 05:19.21 | hypa7ia | whole bunch of things, i don't want to do cisco crap for the rest of my life, going back to school part time, and got a cool gig teaching infosec courses at Big Blue |
| 05:19.28 | Qwell | ahh |
| 05:19.52 | hypa7ia | distance ed + occasional teaching gig = lots of time to do consulting work and stuff |
| 05:20.05 | hypa7ia | overall, they just couldn't match that. |
| 05:20.15 | hypa7ia | they should have hired me on fulltime 3 months ago |
| 05:20.25 | Qwell | oh, you were still a temp? screw that |
| 05:20.28 | hypa7ia | a year at the intern rates they were paying was too much :( |
| 05:21.21 | Qwell | I'm hoping to put some pressure on my boss fairly soon here. He's starting to get really excited about this * project we're working on...and I'm getting calls about employment left and right |
| 05:22.03 | hypa7ia | awesome |
| 05:22.06 | hypa7ia | leverage is good |
| 05:23.55 | graphyx_home | I am getting this notice in asterisk Host '192.168.0.1' does not implement 'REGISTER' |
| 05:24.11 | mog_home | hey stop trying to get on my box graphyx.... |
| 05:24.23 | graphyx_home | I have just setup a new asteirsk and am trying to get an express talk windows softphone to link to it. |
| 05:24.33 | mog_home | i have a firewall |
| 05:24.38 | *** join/#asterisk MrBlack (n=bharatsa@210.211.246.47) |
| 05:24.44 | graphyx_home | any ideas on that? |
| 05:24.45 | MrBlack | hello All |
| 05:25.26 | *** join/#asterisk ahattar (n=kjsd@ool-435292d6.dyn.optonline.net) |
| 05:25.36 | MrBlack | I am trying to configure queues using realtime.. And I am thru with the steps mentioned in the voip-info.org... |
| 05:25.58 | MrBlack | Do I need to make any alterations in the queues.conf file? |
| 05:26.12 | MrBlack | please let me know? |
| 05:29.38 | *** join/#asterisk bmg505 (n=leon@rndf-146-57-40.telkomadsl.co.za) |
| 05:29.40 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 05:32.27 | Flauto | SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead, please tell me what does this mean |
| 05:34.05 | MrBlack | anybody knows configuring the Queues onthe realtime |
| 05:34.09 | MrBlack | please let me know |
| 05:36.20 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 05:36.42 | shmooz | yo |
| 05:37.38 | Flauto | hi |
| 05:38.20 | Math` | Flauto: it tells you to use Set(CALLERID(name)=value) instead |
| 05:38.44 | MrBlack | Shmooz: do you know how to configure the realtime for queuees? |
| 05:39.12 | Math` | will u also individually ask the 300 other users? lol :P |
| 05:39.19 | shmooz | uhm well I did queues not sure about real tim |
| 05:39.20 | tainted_ | what is a SIP RESPONSE of 503? |
| 05:39.30 | Math` | tainted_: service unavailable |
| 05:39.44 | MrBlack | Math: there is nothing to laugh i feel |
| 05:39.52 | MrBlack | I need the answer soon |
| 05:40.02 | tainted_ | how about 400 "Bad Request" |
| 05:40.02 | MrBlack | thats the reason I am asking |
| 05:40.34 | Math` | tainted_: you're getting that with your gateway? |
| 05:40.49 | tainted_ | my provider |
| 05:42.43 | shmooz | MrBlack what do you mean realtime vs what? |
| 05:44.15 | rajiv | damn. this is the problem i am seeing with my tdm400p http://lists.digium.com/pipermail/asterisk-users/2004-July/054356.html |
| 05:45.08 | *** join/#asterisk wolfson (n=ggggg@208.25.254.120) |
| 05:49.07 | Nugget | rajiv: many of us have that problem. |
| 05:49.19 | rajiv | oh? |
| 05:49.27 | rajiv | any solutions besides rebooting the box ? |
| 05:49.33 | Nugget | not that I'm aware of |
| 05:49.47 | ikarus | Hmmmmm, I love predictive dialing |
| 05:49.57 | rajiv | has digium done anything about it ? |
| 05:50.05 | ikarus | saves me having to figure out a dial set on the phones |
| 05:50.42 | *** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net) |
| 05:52.12 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:52.29 | rajiv | Nugget: what do you do about it? |
| 05:52.52 | Nugget | I stopped using the FXO ports on my TDM400P. |
| 05:53.08 | MrBlack | Shmooz:havent you heard of configuring the extensions.conf file and the voicemail.conf using real time? |
| 05:53.31 | rajiv | Nugget: no way. isn't that drastic? i hate not using the hardware i have |
| 05:53.37 | mog_home | rajiv this is an older issue |
| 05:53.43 | Nugget | it's not as drastic as missing calls. |
| 05:53.43 | mog_home | what rev of the card do you have |
| 05:54.10 | rajiv | mog_home: can i get that info in software? |
| 05:54.46 | mog_home | how old is your hw |
| 05:54.58 | rajiv | 2 years maybe |
| 05:54.58 | asterboy | When I go to a computer recycling depot...all I see is * servers. |
| 05:55.05 | Math` | lol |
| 05:55.23 | mog_home | rma your card |
| 05:55.28 | mog_home | it will rock your world |
| 05:55.29 | mog_home | and work |
| 05:55.58 | hypa7ia | asterboy: the sad thing is... most of those would be faster than a Nortel BCM... those suckers are celeron 733's |
| 05:56.39 | rajiv | i got the card in feb/mar 2004 |
| 05:56.49 | mog_home | its all good |
| 05:56.57 | asterboy | lol...that is so where we are...what is sad is that not a lot of people know that. |
| 05:58.27 | jarrod | hey |
| 05:58.29 | rajiv | mog_home: k i'll ask about an rma. not going to be fun. |
| 05:58.38 | rajiv | does digium cross ship ? |
| 05:58.47 | jarrod | how come asterisk manager doesnt show my call events, but when i do a reload it shows the reload event and IAX registry events? |
| 05:59.13 | mog_home | email me directly |
| 05:59.15 | jarrod | the user im using is set for read = system,call,log,verbose,command,agent,user |
| 05:59.17 | mog_home | will make it easy |
| 05:59.22 | mog_home | and we do cross ship |
| 05:59.24 | rajiv | nice |
| 05:59.31 | rajiv | mog_home: query ok? |
| 05:59.39 | mog_home | query? |
| 06:01.16 | jarrod | asterisk manager was showing the events |
| 06:01.20 | jarrod | butnothing changed in manager.conf |
| 06:01.55 | MrBlack | does anybody know about configuring the Asterisk Queues on real time? |
| 06:02.05 | asterboy | Astrisk Emulation running on a multiprocessor...now that would be interesting...VMWare? |
| 06:02.06 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
| 06:02.17 | Qwell | MrBlack: There is a thing on the wiki for it. |
| 06:02.39 | MrBlack | well I am thru with the steps given in the Wiki |
| 06:02.41 | Qwell | lucky you, I happen to have the page up already. http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue |
| 06:02.53 | MrBlack | but need to ask you one more thing |
| 06:02.59 | Qwell | unixodbc |
| 06:03.23 | MrBlack | are there any changes to be made to queues.conf file ? |
| 06:03.28 | Qwell | don't think so |
| 06:03.33 | Qwell | just extconfig.conf |
| 06:03.46 | MrBlack | so should it be a blank file? |
| 06:03.58 | shmooz | well I'll be danged, didn't know realtime |
| 06:04.11 | MrBlack | cos everything is coming from DB |
| 06:04.15 | MrBlack | is it? |
| 06:04.19 | Qwell | MrBlack: I think you still need the general stuff |
| 06:04.20 | shmooz | mysql ? |
| 06:04.26 | Qwell | just not the actual queue definitions |
| 06:04.27 | MrBlack | yes |
| 06:04.40 | shmooz | I can do the stuff without the DB , just conf files |
| 06:04.43 | Qwell | ie; the stuff in [myQueue] would go in the DB |
| 06:05.15 | shmooz | since when has this DB been around ? |
| 06:05.29 | Math` | http://www.mediatrix.com/buy.php <-- register to receive free demo unit :o |
| 06:05.40 | Qwell | demo unit? |
| 06:05.40 | shmooz | I'm gonna have to update my php phone adding panel to do the DB part as well |
| 06:05.54 | MrBlack | is it only the stuff in the [myQueue] would be in the DB or the details in the general context as well? |
| 06:06.07 | Qwell | MrBlack: the stuff in general would still be in the file |
| 06:06.13 | MrBlack | alright |
| 06:06.16 | MrBlack | Qwell |
| 06:06.20 | MrBlack | thanks a lot |
| 06:06.28 | Qwell | MrBlack: paypal.com ;] |
| 06:06.35 | MrBlack | i will try my hands |
| 06:06.36 | MrBlack | :) |
| 06:06.39 | MrBlack | alright |
| 06:07.54 | Qwell | brb |
| 06:08.44 | rajiv | how much are x100m modules? |
| 06:08.50 | Qwell | m? |
| 06:08.54 | Qwell | p? |
| 06:09.03 | mog_home | 100 or less |
| 06:09.06 | shmooz | close to 100 |
| 06:09.07 | mog_home | its been a bit |
| 06:09.10 | drumkilla | mog_home: ! |
| 06:09.16 | mog_home | depends where you go from |
| 06:09.18 | *** join/#asterisk implicit (n=implicit@216.13.124.132) |
| 06:09.36 | shmooz | wasn't there some 56k modem that could works as zaptel card ? |
| 06:09.52 | Qwell | the x100p |
| 06:11.05 | mog_home | half way there |
| 06:11.29 | shmooz | MrBlack voipinfo says 'The easiest way to get existing *.conf files into the database is by using bwk's perl script' |
| 06:11.43 | shmooz | http://www.krisk.org/asterisk/ast2sql.pl |
| 06:12.01 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 06:12.09 | rajiv | Qwell: x100m modules for a tdm400p card |
| 06:12.26 | *** join/#asterisk marc324 (n=marc3234@206-248-132-178.dsl.teksavvy.com) |
| 06:12.29 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 06:13.08 | rajiv | ohh. $75 direct from digium http://store.digium.com/product_view.php?category=1&product_code=SOLOFXS |
| 06:13.23 | rajiv | maybe i'll replace my x100p card with a module and free up a pci slot |
| 06:13.24 | mog_home | wow they got cheaper |
| 06:13.34 | rajiv | would that also free up an interrupt ? |
| 06:13.36 | Math` | hell yeah |
| 06:14.16 | Math` | how are TDM-FXS modules working compared to ATAs? |
| 06:14.19 | rajiv | oops. the x100m fxo modules are $85 |
| 06:15.16 | rajiv | still. $85 for an interrupt... heh |
| 06:16.59 | rajiv | slightly cheaper at voipsupply for the oem versions |
| 06:18.04 | Inv_arp | whos the provider to use for outgoing toll free 800 numbers again? |
| 06:18.21 | Math` | uhm fwd routes some of them |
| 06:18.42 | joelsolanki | right now i m using g729 and alaw ..endpoint is linksys pap2. as alaw consumes around 80 kbps i want to change with g726 as it is free to use. i tried to use that but it is telling me " no compatilbe codecs" in CLI |
| 06:18.59 | joelsolanki | any ideas |
| 06:19.07 | Math` | g726 is royalty-free? |
| 06:19.15 | rajiv | thx all. bbl, trying new kernel.. |
| 06:19.40 | joelsolanki | not free. i think it is passthru |
| 06:19.53 | Math` | g729 is free for passthru too... |
| 06:20.06 | Math` | anything is free for passthru because the codec is not used on asterisk |
| 06:20.11 | Math` | it just forwards the data |
| 06:20.33 | joelsolanki | yes. my problem is that i want to change alaw to some other codec which is free? |
| 06:20.54 | Math` | gsm but the sound quality will drop |
| 06:21.14 | Math` | you should check the list of codecs supported in the Linksys PAP2 |
| 06:22.05 | joelsolanki | Math: let me check in the linksys |
| 06:22.16 | Math` | joelsolanki: how much is a pap2 retailing btw? |
| 06:22.33 | joelsolanki | bandwidth ? |
| 06:23.08 | Math` | how much can you buy a pap2 for? :P |
| 06:23.49 | joelsolanki | it is around 85 to 95 $ |
| 06:24.13 | Math` | ok not bad |
| 06:24.18 | joelsolanki | Math: do u have any experience with sipura 2100 |
| 06:24.45 | *** part/#asterisk litage (n=nick@203.220.55.70) |
| 06:25.08 | Math` | no |
| 06:25.16 | joelsolanki | Math: sipura 2100 has dual g279 codec support. so that might reduce my bandwidth usage. |
| 06:25.34 | Math` | I only played with mediatrix stuff (provider shipped it for free so I used it :P) |
| 06:25.43 | joelsolanki | :) |
| 06:25.51 | Math` | and they have dual g729 too :) |
| 06:26.03 | joelsolanki | mediatrix ..how much it cost ? |
| 06:26.16 | Math` | retail 150$, or on ebay for less than 20$us when there's a deal |
| 06:26.32 | Math` | but they are locked for a provider at that price |
| 06:26.44 | Math` | but I can guide you thru the unlocking procedure |
| 06:26.57 | Math` | look for the mediatrix 2102 model |
| 06:27.09 | Qwell | Whats a mediatrix? |
| 06:27.09 | Math` | 2 port fxs w/ fax support |
| 06:27.16 | Qwell | okay then |
| 06:27.17 | joelsolanki | hmm ok. let have a look for it. |
| 06:27.26 | Math` | Qwell: http://www.mediatrix.com/products_devices.php?prodid=14 |
| 06:28.04 | Qwell | sip? |
| 06:28.13 | Math` | SIP/H323/MGCP (flashable) |
| 06:28.59 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 06:29.22 | Math` | even supports 802.1q vlan tagging heh |
| 06:29.49 | joelsolanki | sounds good. |
| 06:30.07 | joelsolanki | Math: any idea how much bandwidth both lines consumes ? |
| 06:30.19 | Math` | g729 is 8kbps per concurrent call (+ ip overhead) |
| 06:31.04 | *** join/#asterisk P4C0 (i=1000@201.224.107.47) |
| 06:31.08 | joelsolanki | so whats total ? |
| 06:31.16 | joelsolanki | i guess 25 per call. |
| 06:31.31 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 06:32.29 | Math` | 8 per call |
| 06:32.38 | Math` | uhm 16 sorry |
| 06:33.43 | P4C0 | hello guys, I'm planning to have a full PBX setup with asterisk, so I need some software client (windows and linux) and a card for plugging the telephone line, could somebody recomend me a good sip phone software for linux and windows and a good hardware card for that (of course with good interaction with asterisk on linux) |
| 06:33.56 | justinu | anyone know how to implement "connected party identification" on polycom sip? |
| 06:34.09 | Math` | P4C0: x-lite by xten technologies, both win and linux |
| 06:34.21 | justinu | like what SIP RFC/Draft implements that? |
| 06:34.34 | Math` | P4C0: digium sells pci cards depending on your needs... you want to plug 1 regular telephone line? |
| 06:34.41 | IronHelix | P4C0- Digium and Sangoma both make good interface cards |
| 06:34.47 | shmooz | theres a linux version of x-lite ? |
| 06:34.53 | IronHelix | digium tdm400 and tdm2400 series are good if you have analog lines |
| 06:34.54 | Math` | shmooz: yup :) |
| 06:34.56 | Qwell | yep, but it's just as bad as the windows version |
| 06:34.56 | shmooz | I was using wine to run it all that time :\ |
| 06:35.03 | Math` | lol |
| 06:35.08 | P4C0 | Math`yes but if I can get with 2 analog lines that will be nice |
| 06:35.08 | Math` | its new tho |
| 06:35.11 | Math` | maybe 1 month |
| 06:35.11 | shmooz | the win binary runs fine in wine ;) |
| 06:35.12 | IronHelix | you can also use any SIP or IAX based softphone |
| 06:35.18 | Qwell | Math`: xlite for linux? 3-4 at least |
| 06:35.23 | Math` | oh |
| 06:35.50 | Qwell | it's not that it's poorly written... |
| 06:35.50 | Math` | P4C0: uhm you want to use your actual analog lines? or you want analog lines out of the pbx? |
| 06:35.50 | Qwell | it's...poorly designed |
| 06:36.01 | IronHelix | P4C0- check out the digium tdm400 series, its a 4 port analog card and each port can be either fxo (connect to line) or fxs (connect to phone) based on what modules you plug in it |
| 06:36.02 | shmooz | looks nice tho |
| 06:36.31 | Math` | Qwell: what do you suggest as a free sip softphone then? |
| 06:36.37 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 06:36.45 | justinu | he suggests you buy eyebeam |
| 06:37.01 | shmooz | there was some kde softphone |
| 06:37.02 | P4C0 | Math`: no I want to use the lines for input, for output it will all be ethernet... or maybe a physical sip phone |
| 06:37.16 | Qwell | Math`: I haven't found anything yet |
| 06:37.21 | P4C0 | is x-lite opensource? |
| 06:37.25 | Math` | no |
| 06:37.25 | IronHelix | no |
| 06:37.47 | P4C0 | anyone opensource? |
| 06:37.49 | shmooz | decompile and it will be |
| 06:37.53 | Math` | lol |
| 06:38.03 | Math` | open-source pure assembly sip softphone! |
| 06:38.09 | P4C0 | shmooz:sure, but some people also have a life too |
| 06:38.13 | IronHelix | http://www.voip-info.org/wiki-Open+Source+VOIP+Software |
| 06:38.13 | Math` | with precompiled binaries available |
| 06:38.20 | shmooz | pure assembly is for 386 486 days |
| 06:38.42 | justinu | i can't believe there's no #sip |
| 06:39.31 | P4C0 | justinu: well it's a protocol... is like #rtp #ftp #http and so... usually there's no channel for that :p |
| 06:39.42 | justinu | i know, but i need to talk about that protocol |
| 06:40.10 | shmooz | so talk here I guess |
| 06:40.15 | P4C0 | yep |
| 06:40.17 | Math` | /join #sip,#rtp,#h323,#mgcp |
| 06:40.18 | IronHelix | theres a #voip but only 4 ppl in it |
| 06:40.21 | shmooz | 600 eyeballs lookin |
| 06:40.22 | P4C0 | or contact the authors :p |
| 06:40.22 | Math` | damn |
| 06:40.31 | justinu | i'm trying to implement the "connected party identification" feature on the polycom 501 |
| 06:40.36 | Qwell | /j #rfc3389 |
| 06:40.59 | justinu | now, the polycom spec says it does it, but it gives no specifics on how it's implemented |
| 06:41.21 | Qwell | justinu: that like presence? |
| 06:41.24 | justinu | connected party show you the ultimate destination of the call |
| 06:41.30 | Math` | oh |
| 06:41.35 | justinu | so, if user B forwards call to C |
| 06:41.43 | justinu | a calls B, but sees he's connected to C via the display |
| 06:42.11 | justinu | i'm trying to figure out which rfc/draft that's discussed in |
| 06:42.35 | Qwell | rpid? |
| 06:42.49 | justinu | that's like only in invites tho, i think |
| 06:42.57 | Qwell | no clue |
| 06:43.10 | justinu | it has to do with the 302 moved temporarily being sent |
| 06:43.25 | justinu | you see what the target of the 302 forward is, and display it on the phone |
| 06:44.18 | justinu | if a calee forwards his phone to an outside number, the caller will see it on the display |
| 06:44.22 | justinu | i think that's a kick ass feature |
| 06:44.38 | Math` | it is |
| 06:44.51 | justinu | Math`: help me get it working :) |
| 06:45.09 | Math` | I'd love too, but I aint have any polycoms :P |
| 06:45.14 | justinu | hmm |
| 06:45.18 | Math` | ship me one |
| 06:45.21 | Qwell | I'm sure he'd be glad to mail you two |
| 06:45.21 | Math` | and I'll help you :P |
| 06:45.33 | Math` | pm me for shipping info |
| 06:45.35 | justinu | shaking me down already |
| 06:45.45 | Qwell | while you're at it, I can help too. |
| 06:45.49 | Math` | lol |
| 06:45.52 | Math` | he asked me! |
| 06:46.01 | shmooz | no me first me first!! |
| 06:46.11 | Math` | but if we need to test transfer we need 3 partys so Qwell can have 1 too |
| 06:46.18 | justinu | lol, i'm off guys... zzzzzzzzzzzzzzzzz |
| 06:46.22 | Qwell | yep |
| 06:46.24 | Dr_Ray | and people say this channel is not helpfull |
| 06:46.38 | shmooz | moody aren't we |
| 06:46.58 | shmooz | or he was just talkin crap like everyone else |
| 06:47.44 | P4C0 | I'm on http://store.digium.com if I want to plug a regular telephone line into my pc (for input) which card should I buy? digital interface cards? (they seem to be the only cards) |
| 06:48.06 | Qwell | P4C0: you want analog cards. fxo to be specific |
| 06:48.47 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 06:49.00 | P4C0 | Qwell, fxo? |
| 06:49.02 | Qwell | fxo |
| 06:49.31 | IronHelix | ~fxofxs |
| 06:49.34 | jbot | somebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 06:49.34 | Qwell | bot don't like you |
| 06:49.36 | P4C0 | what is fxs? |
| 06:50.01 | P4C0 | oks :) |
| 06:50.23 | P4C0 | when they say module they mean pci card? |
| 06:50.26 | IronHelix | no |
| 06:50.39 | IronHelix | the digium tdm400 is a pci card |
| 06:50.46 | IronHelix | it has space for up to 4 daughterboards |
| 06:50.59 | asterboy | and mix and match. |
| 06:51.00 | IronHelix | you can plug in any mix of FXO (red) and FXS (green) daughterboards |
| 06:51.09 | asterboy | I'm looking for an S100M FXS. |
| 06:51.10 | P4C0 | http://store.digium.com/product_view.php?category=1&product_code=RTDM11B <--- what about this one? |
| 06:51.30 | Qwell | P4C0: sure, if you need an fxo and an fxs |
| 06:51.45 | Math` | P4C0: you can buy the card only from a digium reseller |
| 06:51.49 | IronHelix | thats a tdm400 board wtih two fxo and two fxs ports |
| 06:51.57 | IronHelix | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 06:52.01 | IronHelix | this is a better view of the card itself |
| 06:52.08 | P4C0 | no I only need a fxo |
| 06:52.10 | MrBlack | Hello All |
| 06:52.13 | Qwell | P4C0: how many? |
| 06:52.13 | IronHelix | then you can save money |
| 06:52.25 | IronHelix | the modules are about $75 each and you can buy a card with only the modules you need |
| 06:52.28 | IronHelix | and add more later if you need to |
| 06:52.38 | MrBlack | I am looking for logging the voicemail in multiple voicemail boxess, |
| 06:52.40 | P4C0 | yes, but I'll like to have the posibility to add 2 lines, but I only have one pci free |
| 06:52.45 | Qwell | wow, digium stopped selling small bundles |
| 06:52.58 | P4C0 | IronHelix: cool, thanks |
| 06:53.00 | Qwell | P4C0: each "card" only takes one slot. one "card" can have 4 modules |
| 06:53.07 | Math` | yeah I see that |
| 06:53.09 | MrBlack | but its loggig the voicemail only in the first voiceemail box specified.. |
| 06:53.11 | IronHelix | p4c0 try this link http://www.voipsupply.com/index.php?cPath=99_103 |
| 06:53.18 | IronHelix | they have the tdm card with any configuration of modules |
| 06:53.24 | IronHelix | keep in mind the pictures are wrong tho |
| 06:53.26 | MrBlack | does anybody know the solutoin to this |
| 06:53.32 | IronHelix | just look at descriptions and prices |
| 06:53.42 | Math` | Qwell: the thing is... ATAs are a LOT cheaper than TDM cards, and are often more practical to install |
| 06:53.46 | P4C0 | thanks for the link IronHelix |
| 06:53.55 | IronHelix | if you want two FXO, try this- http://www.voipsupply.com/product_info.php?products_id=292 |
| 06:53.58 | Math` | but for FXO TDM might be cheaper |
| 06:54.24 | IronHelix | it has two fxo (red) modules and the card. You will then have space to add two more later if you want |
| 06:54.24 | MrBlack | Qwell: do you knw about logging the voicemail in multiple voice mail boxes? |
| 06:54.45 | Qwell | MrBlack: I think you can call Voicemail(100&101) |
| 06:54.49 | Qwell | I believe that will leave the message for extension 100 and 101 |
| 06:54.49 | MrBlack | no it isnt working |
| 06:54.54 | Qwell | cvs head? |
| 06:55.05 | Qwell | think I saw a bug about that recently |
| 06:55.26 | MrBlack | Asterisk built by root@localhost.localdomain on a i686 running Linux on 2005-11-14 07:47:02 UTC |
| 06:55.32 | MrBlack | does that matter |
| 06:55.35 | Math` | no |
| 06:55.46 | Qwell | MrBlack: When did you download it from cvs? |
| 06:56.01 | Qwell | I think that bug may have been closed |
| 06:56.01 | MrBlack | just a few days back |
| 06:56.05 | Math` | cvs update |
| 06:56.13 | LostFrog | wow.. multiple voicemail boxes? |
| 06:56.13 | Math` | a lot of changes happened in the past week |
| 06:56.13 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-82-80-162-23.red.bezeqint.net) |
| 06:56.26 | Math` | LostFrog: well... its just writing a file more than once |
| 06:56.50 | Qwell | http://bugs.digium.com/view.php?id=5704 |
| 06:56.52 | Qwell | closed |
| 06:57.08 | Qwell | MrBlack: run a `make update` from your asterisk source dir, then run a make install |
| 06:57.44 | LostFrog | Math`: still not a concept I would have though of. |
| 06:58.22 | Math` | :) |
| 06:58.23 | MrBlack | Qwell:which versions of Asterisk support multiple voicemail boxess? |
| 06:58.27 | Qwell | got me |
| 06:58.32 | Qwell | I think it's been there a while |
| 06:58.46 | Qwell | cvs and 1.2 do - that's all that matters anymore |
| 06:58.53 | IronHelix | 1.0.x does too |
| 06:58.58 | IronHelix | i've used that feature for a while |
| 06:59.08 | IronHelix | whiel = months |
| 06:59.15 | Qwell | nobody uses 1.0.x anymore :P |
| 07:00.11 | MrBlack | anything need to be configured to setup multiple voicemail box, other than specifying one more mail box number for the peer. |
| 07:00.17 | MrBlack | Qwell? |
| 07:00.23 | Qwell | huh? |
| 07:00.24 | drumkilla | Qwell: it matters!! |
| 07:00.26 | drumkilla | :) |
| 07:00.34 | Qwell | drumkilla: to you... :p |
| 07:00.38 | Qwell | and only for a few more days/weeks, heh |
| 07:00.41 | drumkilla | you have to admit that Asterisk 1.0 is a very well known platform at this point |
| 07:00.46 | Qwell | well, yeah |
| 07:00.58 | drumkilla | we know about the issues, and know how to work around them |
| 07:01.10 | Qwell | "upgrade to cvs head" |
| 07:01.11 | drumkilla | the way it works hasn't changed in over a year |
| 07:01.22 | Qwell | yeah, 1.0 is good |
| 07:01.23 | drumkilla | so that's pretty nice for a lot of people |
| 07:01.58 | Qwell | so, you're gonna be doing 1.2 also now? |
| 07:02.02 | drumkilla | yup |
| 07:02.09 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 07:02.09 | Qwell | I know there were talks about it before |
| 07:02.19 | drumkilla | well, it's official now, heh |
| 07:02.25 | Qwell | cool |
| 07:02.40 | drumkilla | and Digium will be sponsoring my work |
| 07:02.49 | Qwell | russell->funlevel--; everybody_else->funlevel++; |
| 07:02.54 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:03.02 | P4C0 | one little question, does asterisk allows me to configure one regular phone (connected into a fxs outlet) to work as the main control unit of the PBX? (basically all calls get to that ext, and the person there forwards them?) |
| 07:03.02 | Qwell | sponsoring, as in paying you? :) |
| 07:03.06 | drumkilla | yes |
| 07:03.08 | Qwell | nice |
| 07:03.12 | Qwell | abuot time. ;] |
| 07:03.21 | drumkilla | yeah, so that helps the funlevel :) |
| 07:03.29 | Qwell | --funlevel++ |
| 07:03.34 | drumkilla | lol ... |
| 07:03.35 | Qwell | (hmm...does that work?) |
| 07:03.38 | Math` | lol |
| 07:03.42 | Math` | no |
| 07:03.44 | Math` | not in C |
| 07:03.47 | Qwell | why not? |
| 07:03.50 | Qwell | oh, heh |
| 07:03.58 | Math` | because --funlevel returns a value |
| 07:04.01 | Math` | and you cant do 1++ |
| 07:04.07 | Qwell | oh |
| 07:04.10 | Math` | in c++ you can do (--funlevel)++ |
| 07:04.21 | drumkilla | indeed |
| 07:04.23 | Math` | or --(funlevel++) |
| 07:04.24 | drumkilla | silly c++ |
| 07:04.27 | Math` | lol |
| 07:04.30 | Qwell | --c++ |
| 07:04.30 | hypa7ia | python doesn't have ++ and -- |
| 07:04.35 | hypa7ia | this makes me sad |
| 07:04.37 | Qwell | is THAT valid? |
| 07:04.44 | Math` | Qwell: no, need ( ) |
| 07:04.51 | Qwell | since "c++" is the var name? |
| 07:05.00 | LostFrog | In c++ operators return a reference. |
| 07:05.02 | Math` | invalid char '+' for identifier :P |
| 07:05.36 | shmooz | c = c + 1 |
| 07:05.52 | shmooz | goto 10 |
| 07:05.57 | Qwell | --c = c++;? |
| 07:06.01 | Qwell | Would THAT work? |
| 07:06.08 | shmooz | not logical |
| 07:06.09 | drumkilla | just "c." would be valid in prolog. |
| 07:06.10 | LostFrog | Only in C++, Qwell. |
| 07:06.39 | infinity1 | the functionality that replaces dbput dbdel from using a flat file database and to use mysql is called what? |
| 07:06.52 | infinity1 | i'm trying to figure out what i should ask google for :) |
| 07:07.05 | shmooz | ask it that |
| 07:07.07 | Math` | Qwell: yeah it works lol |
| 07:07.09 | drumkilla | you mean sqllite? |
| 07:07.13 | Qwell | sweet |
| 07:07.24 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 07:07.25 | Qwell | now...what's it do? |
| 07:07.31 | Math` | lol |
| 07:07.33 | Math` | nothing |
| 07:07.41 | shmooz | defies logic |
| 07:07.41 | Math` | c = c - 1 + 1; // that |
| 07:07.46 | infinity1 | shmooz: err ..thats not a search engine friendly keyword |
| 07:07.51 | Math` | bah I tested it, c was 0 and is still 0 |
| 07:08.22 | shmooz | get the asm with -s |
| 07:08.26 | shmooz | or was it -S |
| 07:08.28 | Math` | -S |
| 07:08.31 | shmooz | yeah |
| 07:08.32 | LostFrog | Makes sense, math. |
| 07:08.42 | Math` | if course it does |
| 07:09.01 | Math` | why do I get pthread stuff in that lol |
| 07:09.22 | P4C0 | bye bye guys, thanks |
| 07:09.26 | Math` | np |
| 07:09.29 | Math` | too late |
| 07:09.34 | Math` | nite drumkilla |
| 07:09.44 | drumkilla | g'night |
| 07:10.59 | Math` | it seems to optimize it |
| 07:11.14 | infinity1 | after googling around voip info ..it appears you can't replace dbput/del with mysql |
| 07:11.43 | LostFrog | --c = c++ is the same as --c |
| 07:11.50 | LostFrog | or c-- for that matter. |
| 07:12.07 | Math` | no |
| 07:12.10 | LostFrog | yes,' |
| 07:12.15 | Math` | well, test it |
| 07:12.26 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 07:12.27 | LostFrog | I just did. |
| 07:12.39 | shmooz | can't be |
| 07:12.42 | pooh_ | morn |
| 07:12.59 | shmooz | thats probably an error code |
| 07:13.01 | Math` | LostFrog: well not for g++ |
| 07:13.14 | Math[laptop] | <PROTECTED> |
| 07:13.14 | Math[laptop] | <PROTECTED> |
| 07:13.14 | Math[laptop] | <PROTECTED> |
| 07:13.23 | Math[laptop] | laptop:~# ./test |
| 07:13.23 | Math[laptop] | c is: 1 |
| 07:13.56 | LostFrog | <PROTECTED> |
| 07:13.56 | LostFrog | <PROTECTED> |
| 07:13.56 | LostFrog | <PROTECTED> |
| 07:13.57 | LostFrog | <PROTECTED> |
| 07:14.09 | LostFrog | 10 |
| 07:14.09 | LostFrog | 9 |
| 07:14.21 | Math[laptop] | gcc version 4.0.3 20051023 (prerelease) (Debian 4.0.2-3) |
| 07:14.40 | Qwell | rofl |
| 07:14.40 | LostFrog | gcc version 3.3.5 (Debian 1:3.3.5-13) |
| 07:14.43 | LostFrog | weird. |
| 07:15.14 | Qwell | wait |
| 07:15.19 | Math` | behavior differs in compiler versions heh |
| 07:15.25 | Qwell | Math[laptop]: in yours, what happens if you set c to something higher than 1? |
| 07:15.35 | Math` | I've set c = 10 and it gives 10 too |
| 07:15.39 | Qwell | funky |
| 07:16.07 | LostFrog | It is probably not defined behavior in Stroustrup. |
| 07:16.45 | shmooz | did C++0x get released? |
| 07:17.11 | shmooz | anyway I should be askin google.. |
| 07:17.19 | Math` | probably :) |
| 07:17.30 | Math` | I'll ask google the same |
| 07:17.43 | LostFrog | I think I'm going to ask the back of my eyelids. |
| 07:17.47 | LostFrog | Night, all. |
| 07:18.05 | Math` | nite |
| 07:18.24 | Qwell | I'm getting the same behavior as LostFrog |
| 07:18.33 | Math` | you probably have the same g++ version |
| 07:18.44 | Qwell | yeah, 3.3.5 |
| 07:18.54 | Math` | logical you get the same |
| 07:19.30 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 07:19.53 | Qwell | --c = ++c is 10, and --c = c++ is 9 |
| 07:20.14 | Math` | uhu |
| 07:20.43 | Math` | both are 10 |
| 07:20.47 | Qwell | heh |
| 07:23.45 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
| 07:23.47 | xbmodder_lappy | hi |
| 07:24.28 | Math` | damn astbill, I get the *same* mysql version they ask for and the damn .sql has errors |
| 07:25.29 | konfuzed | so that edubuntu install is done |
| 07:25.40 | konfuzed | moodle is installed and so is asterisk |
| 07:25.52 | konfuzed | that was way simpler than debian cvs |
| 07:25.54 | konfuzed | ;^) |
| 07:26.01 | Math` | ? |
| 07:26.05 | konfuzed | true I still have to configure it I know |
| 07:26.08 | Qwell | Math`: having somebody test on 2.95 and 3.4, heh |
| 07:27.06 | Math` | lol |
| 07:27.46 | xbmodder_lappy | <PROTECTED> |
| 07:28.43 | Math` | who do who think |
| 07:28.51 | Qwell | 3.3.6=9 |
| 07:30.16 | Qwell | "<awol> Qwell: the 3.4 == 9 and the 2.95 == 10" |
| 07:30.22 | Math` | lol |
| 07:30.30 | Math` | 4.1 will == 9 I guess |
| 07:31.23 | Qwell | so, what does this prove? |
| 07:31.23 | Qwell | gcc 3.x cannot be trusted. :P |
| 07:31.33 | Math` | hehe |
| 07:33.06 | shmooz | hey konfuzed !!! |
| 07:33.11 | shmooz | wazup |
| 07:33.29 | shmooz | good stuff |
| 07:34.25 | shmooz | Qwell some things still depend on GCC 3.x tho no? |
| 07:34.42 | Qwell | shmooz: of course |
| 07:34.43 | Math` | ah fixed astbill.sql, they were using ";;" as delimiter, so the binary thinks its the end of the instruction, and ignored it |
| 07:35.31 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 07:36.01 | *** join/#asterisk skopii (n=john@dsl017-073-026.chi4.dsl.speakeasy.net) |
| 07:36.05 | skopii | oi oi |
| 07:36.26 | skopii | I am wondering if anyone wants to help a noob in need =] |
| 07:36.33 | Math` | just ask |
| 07:36.36 | skopii | I want to setup asterisk at the office |
| 07:36.37 | Math` | we do that every day |
| 07:36.45 | IronHelix | its what we're here for :) |
| 07:36.46 | skopii | I am confused about how SIP works |
| 07:36.49 | Math` | (helping noobs, not setting up * in offices, :P) |
| 07:36.57 | skopii | haven't tackled the RFC yet |
| 07:37.00 | skopii | but a few questions |
| 07:37.02 | IronHelix | well we can set up * in offices but you generally have to pay us for that |
| 07:37.04 | skopii | we have a call center |
| 07:37.09 | Math` | IronHelix: :) |
| 07:37.22 | skopii | and I am wondering if a DID->ser->asterisk is all we need? |
| 07:37.31 | IronHelix | you may not even need SER |
| 07:37.35 | skopii | I mean for incoming calls do we need PSTN termination? |
| 07:37.37 | skopii | hmm |
| 07:37.46 | skopii | well see that brings me to my next question |
| 07:37.47 | Math` | skopii: PSTN *termination* is for *outgoing* calls only |
| 07:37.47 | wasim | skopii: if your incoming calls come on a PSTN, then yes |
| 07:37.54 | IronHelix | dont overcomplicate things if you can avoid it |
| 07:38.00 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 07:38.05 | skopii | but it's possible to get a DID->sip->* |
| 07:38.06 | lme | hello guys |
| 07:38.10 | Math` | skopii: yes |
| 07:38.12 | wasim | skopii: absolutely |
| 07:38.12 | IronHelix | sko- sure it is |
| 07:38.17 | Math` | thats what I have here :) |
| 07:38.18 | skopii | yeah I am horrible at not overcomplicating things heh |
| 07:38.19 | IronHelix | there are a bunch of companies that provide such a service |
| 07:38.26 | wasim | like nufone |
| 07:38.27 | skopii | alright, I saw a few on the wiki |
| 07:38.34 | IronHelix | few- try hundreds |
| 07:38.36 | Math` | skopii: what do you want to setup exactly? |
| 07:38.55 | IronHelix | yes.. tell us |
| 07:38.59 | IronHelix | then we will take your soul |
| 07:39.02 | IronHelix | feed it to mark spencer |
| 07:39.06 | IronHelix | and give you a plan in return! |
| 07:39.13 | Math` | lol |
| 07:39.57 | skopii | I don't know if the sip phones talk to asterisk or SER directly. But I want to have ser route calls to the next available asterisk box, where the dial plan would be identical on each asterisk box, and then those boxes would forward to a different asterisk box where callers would sit on hold |
| 07:40.11 | skopii | I am working on a diagram |
| 07:40.18 | skopii | probably should have finished it before I said whatsup |
| 07:40.30 | IronHelix | ok 1. you dont need more than 1 asterisk box unless you have 1000's of users or are doing failover/highavailability |
| 07:40.31 | Math` | why do you absolutely want ser? |
| 07:40.55 | skopii | I have read that it helps asterisk to scale |
| 07:41.06 | skopii | and I have a bit of experiance w/ LVS/HA |
| 07:41.06 | IronHelix | how far are you scaling? |
| 07:41.41 | skopii | I want the system to handle more than 500 calls (more than we would ever hopefulyl take) |
| 07:41.49 | skopii | and I would like to use commodity hardware |
| 07:42.18 | skopii | ie 'toasters' (P4 2.4GHZ 1GB RAM IDE) |
| 07:42.49 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 07:43.05 | IronHelix | http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning this has some info on how much HW you will need |
| 07:43.11 | Math` | you're aware that even with g729, a high compression codec, you are going to need 4mbps full duplex to handle all that traffic? |
| 07:43.21 | nick125 | hrm, on average, how many kbps does a SIP channel using GSM use? |
| 07:43.55 | Math` | skopii: how many agents will there be in the call center? |
| 07:44.24 | Math` | and do you require call recording, etc...? |
| 07:44.30 | skopii | see thats the thing, I would like to give us room to expand. but right now only ~15 |
| 07:44.49 | IronHelix | ok if you buy HW for 500 users you will way overspend |
| 07:44.55 | nick125 | true.. |
| 07:44.56 | IronHelix | the nice thing with * is its easy to add capacity |
| 07:45.00 | nick125 | maybe 150 max imho |
| 07:45.07 | skopii | what I wanted to do was have asterisk forward calls destined for certian (pools?) to different asterisk boxes |
| 07:45.18 | IronHelix | 1 server can happily do 150 calls if theres no transcoding |
| 07:45.24 | IronHelix | and can do it with transcoding if you have a good CPU |
| 07:45.37 | skopii | so do I not need to consider incoming calls? |
| 07:45.37 | Math` | IronHelix: more than that with SIP reinvites |
| 07:45.43 | nick125 | dont make things more complex then they need to, skopii... |
| 07:45.50 | IronHelix | to nick you listen |
| 07:45.53 | Math` | skopii: a single server is WAY enough to start with |
| 07:46.02 | Math` | if you experiment problem, just add another one, its easy |
| 07:46.15 | IronHelix | if you have 15 people, for now buy a single robust intel or amd based server, get some bandwidth and some IP phones |
| 07:46.22 | IronHelix | you need ONE * box |
| 07:46.35 | nick125 | yeah, thats the thing i like about asterisk is that it is so scalable.. |
| 07:46.44 | skopii | see I don't like the idea of a single point of failure. |
| 07:46.48 | *** join/#asterisk rLg (i=rLg@202.61.49.31) |
| 07:46.48 | IronHelix | ah |
| 07:46.54 | asterboy | so buy two |
| 07:46.56 | Qwell | still not understanding how one would scale queues and such... |
| 07:46.56 | skopii | but I guess I could easily setup a failover asterisk box |
| 07:46.56 | IronHelix | then get two * boxes which mirror each other |
| 07:47.03 | skopii | yeah |
| 07:47.04 | IronHelix | exactly |
| 07:47.10 | skopii | so is SER no good? |
| 07:47.15 | skopii | or do I really just not need it? |
| 07:47.17 | Qwell | SER has its uses |
| 07:47.18 | IronHelix | nothing wrong with SER, you just dont need it |
| 07:47.21 | Math` | skopii: you don't need it |
| 07:47.37 | skopii | is it for providers? |
| 07:47.47 | shmooz | is asterisk as optimised as SER now for sip ? |
| 07:47.57 | IronHelix | as you learn about * and its uses you will better understand how to scale * when your business grows, skopii |
| 07:48.01 | Qwell | SER doesn't do much with SIP...just moves calls |
| 07:48.04 | IronHelix | for now dont overcomplicate things |
| 07:48.11 | Qwell | it isn't a b2bua or anything |
| 07:48.18 | Qwell | ~b2bua |
| 07:48.19 | jbot | methinks b2bua is a back 2 back user agent |
| 07:48.54 | Math` | skopii: the number of phone calls asterisk can handle, if all your phones are SIP and are supporting SIP reinvites, is almost infinite |
| 07:49.25 | IronHelix | you are going to be limited much more by your bandwidth than by your asterisk box |
| 07:49.36 | skopii | heh BW isn't a problem ;] |
| 07:49.46 | Qwell | That's what they all say |
| 07:49.51 | skopii | I work for a webhosting company |
| 07:50.16 | nick125 | that reminds me, how much b/w does (on average) a gsm sip connection use? |
| 07:50.19 | IronHelix | yes but do you work AT the web hosting company? if you work in the datacenter you're all set then, but if you're somewhere else it may be a problem |
| 07:50.26 | IronHelix | nick- about 30k? |
| 07:50.29 | Math` | lol |
| 07:50.40 | nick125 | 30kbps? |
| 07:50.44 | skopii | well thats the plan we are going to setup the asterisk box[es] at the DC |
| 07:50.56 | skopii | then have the office connect to the server |
| 07:50.57 | Math` | skopii: are all the phones at the same location? |
| 07:50.59 | skopii | or does that not work? |
| 07:51.01 | IronHelix | yeah 30kbit/sec... 15kbit/sec times two |
| 07:51.14 | Math` | skopii: well, calls have to get thru the phones, you need bandwidth where the phone is |
| 07:51.20 | nick125 | IronHelix: thats what i noticed in my testing..about 30-40kbps per call |
| 07:51.36 | Math` | skopii: but for now don't worry, 15 calls is not much bandwidth |
| 07:51.40 | skopii | Math`, snap! that would make sense |
| 07:51.54 | IronHelix | if you can put the * server in the datacenter, and the phones are in the same building, you are ok |
| 07:52.04 | IronHelix | but if the phones are somewhere else it doesnt really matter WHERE the * is |
| 07:52.12 | IronHelix | because your problem is getting voice data to your phones |
| 07:52.23 | Math` | yeah thats waht I noticed him |
| 07:53.23 | skopii | it's possible to forward from the asterisk box at the DC to one at the office though right? |
| 07:53.34 | IronHelix | sure |
| 07:53.38 | IronHelix | setup an iax2 trunk |
| 07:54.35 | IronHelix | so you will have (provider) -- Internet --sip-- datacenter * box --iax2-- callcenter * box --sip-- phones |
| 07:54.35 | Math` | iax is the inter-asterisk exchange protocol |
| 07:55.07 | Math` | IronHelix: if everything is SIP, I'd setup sip calls all the way, so reinvites can work between the phones and the provider |
| 07:55.12 | Math` | thus lowering the load on the server |
| 07:55.24 | IronHelix | yeah i was actually just about to type that |
| 07:55.25 | IronHelix | heh |
| 07:55.49 | IronHelix | that way sko while the call will go through that way, the actual voice data once the call is setup will just go phone -> provider |
| 07:55.55 | IronHelix | skips * entirely |
| 07:56.17 | Math` | except... skopii... do you require call recording |
| 07:56.31 | skopii | voicemail? or monitoring and the like? |
| 07:56.36 | IronHelix | monitoring |
| 07:56.36 | Math` | monitoring |
| 07:56.56 | skopii | I don't see why we would...sneaky mgmt might say otherwise but I don't think so |
| 07:57.23 | IronHelix | monitoring breaks reinvite (thing i said above). * needs the audio data to go through it to record calls |
| 07:58.09 | asterboy | monitoring: "So did you see that new girl in the office?", "ya...she's a hottie, I'd like to..." :-> |
| 07:59.03 | IronHelix | that would be one way to screw with employees, every day the system would record one random phone call and email 5 minutes of it to everybody in the department |
| 07:59.23 | asterboy | that would tune them in pretty quick. |
| 07:59.25 | nick125 | lol |
| 08:00.04 | Qwell | send 30 seconds. That way things can be taken out of context very easily |
| 08:00.23 | IronHelix | hehe |
| 08:00.30 | nick125 | lol |
| 08:00.54 | skopii | how expensive is a 1800 DID? |
| 08:01.01 | IronHelix | not very |
| 08:01.01 | Qwell | not very |
| 08:01.10 | nick125 | depends on where you get it from |
| 08:01.11 | Qwell | 2c/min incoming is decent |
| 08:01.11 | *** join/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 08:01.19 | IronHelix | under $50 to get it setup and a few cents / minute |
| 08:01.30 | nick125 | thats about average, is 2c per minute |
| 08:02.09 | Qwell | generally about $50 (one time) for a vanity DID |
| 08:02.38 | Math` | Qwell: 2c/min? what provider |
| 08:02.45 | Qwell | Math`: nufone and asterlink |
| 08:03.17 | nick125 | i wonder if nufone accepts paypal.. |
| 08:03.22 | Qwell | they used to |
| 08:03.32 | Qwell | might still |
| 08:05.05 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 08:06.13 | Math` | nufone charges 25$ if you choose your 1800 number |
| 08:06.13 | Qwell | if |
| 08:06.13 | Qwell | free otherwise |
| 08:06.15 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:09.51 | shido6 | yes we do |
| 08:10.20 | Math` | shido6: your part of nufone? |
| 08:10.33 | shido6 | yes |
| 08:10.44 | Math` | whats the monthly fee for a 1800 did? same as a local one? |
| 08:10.53 | shido6 | $2.50/mo very soon |
| 08:11.02 | Qwell | hmm |
| 08:11.05 | Qwell | that sucks. :( |
| 08:11.15 | Math` | why so? |
| 08:11.39 | shido6 | when you see the new features |
| 08:11.42 | shido6 | you'll see why |
| 08:11.43 | shido6 | :) |
| 08:11.45 | Math` | shido6: are your 1800 did reachable from all area codes? |
| 08:11.48 | Qwell | I only use 60c/mo, heh |
| 08:12.26 | shido6 | in the US48 yes |
| 08:12.30 | Math` | in canada? |
| 08:12.51 | shido6 | no, that isn't the us48 |
| 08:12.56 | Math` | crap :( |
| 08:13.02 | shido6 | did u need a canadian # ? |
| 08:13.11 | Qwell | You'd think it would "Just Work" from anywhere in NA |
| 08:13.25 | Math` | I'd like :P |
| 08:13.38 | Qwell | can nufone do canada tollfree yet? |
| 08:13.40 | syle | i can do US and canada 2 dollars a month |
| 08:13.44 | shido6 | it can but we increase ALL incoming calls to that 8xx number for your account 10 cents/minute |
| 08:13.49 | shido6 | -for your account |
| 08:13.54 | Qwell | y0wza |
| 08:13.55 | Math` | 10 cents/min? no thanks |
| 08:14.01 | Math` | I can get 4c/min from unlimitel |
| 08:14.16 | shido6 | which is why we dont advertise canadian 8xx #'s |
| 08:14.30 | Math` | ok |
| 08:14.42 | Math` | (+2.50$cad/month) |
| 08:14.54 | shido6 | hopefully they have the documentation that states they cna resell 8xx #'s |
| 08:15.17 | shido6 | otherwise they can have their numbers pulled at a moments notice |
| 08:15.39 | *** join/#asterisk TMirage (n=mirage@cust.12.229.adsl.cistron.nl) |
| 08:15.40 | IronHelix | that would suck |
| 08:15.49 | Qwell | IronHelix: it's happened to a few providers, afaik |
| 08:16.02 | Math` | what? you get a number and someone steals it? |
| 08:16.10 | Qwell | no, they just get pulled |
| 08:16.17 | IronHelix | what happens to the companies using DIDs thru the service? do they get their DID bac? |
| 08:16.49 | Qwell | IronHelix: thats what getting pulled means. the provider AND the customer lose the DID |
| 08:17.41 | IronHelix | damn, that could wipe out a business overnight |
| 08:17.50 | shido6 | yeah there are a lot of fly by night companies out there |
| 08:18.20 | IronHelix | i mean a co using a did... if they invest enough in advertising and their biz is primarily phone based |
| 08:18.26 | IronHelix | lose the did = lose the company |
| 08:18.31 | shido6 | that happens |
| 08:19.05 | shido6 | either do the research on teh company you sign up with or risk your life line to the world |
| 08:19.28 | shido6 | and if spielling had anything to do wiff my serwices I tould be up a creAk |
| 08:19.35 | IronHelix | lol |
| 08:19.41 | Math` | lol |
| 08:20.09 | nick125 | lol |
| 08:20.32 | Rawplayer | when you use asterisk over an vpn connection with about 20 to 30 people how much ram do i need in my machine? |
| 08:21.02 | shido6 | ppl have been slapping gig and 2 gigs of ram in those kinds of boxes |
| 08:21.06 | shido6 | and only use 400 of it |
| 08:21.16 | Qwell | yeah, don't need a whole lot |
| 08:21.21 | Rawplayer | hence that openvpn will also run on that box |
| 08:21.29 | shido6 | its Linux |
| 08:21.31 | *** join/#asterisk NirS (n=nirs@84.94.193.142.cable.012.net.il) |
| 08:21.32 | Math` | ah cmon server boards supports up to 48gb of ram, you gotta take advantage of that :P |
| 08:21.34 | NirS | hello |
| 08:21.37 | NirS | anybody home ? |
| 08:21.41 | Qwell | nope |
| 08:21.41 | Math` | no |
| 08:21.45 | Rawplayer | i'am at work |
| 08:21.50 | IronHelix | you'll need some CPU to deal with the VPN encryption, but you wont need a huge pile of RAM |
| 08:21.51 | Rawplayer | well i think its work |
| 08:21.54 | IronHelix | nothing too fancy |
| 08:21.59 | shido6 | but you wont with that setup |
| 08:22.02 | Math` | Im at home but I work home so Im not at home, I'm at work |
| 08:22.03 | NirS | anyone have experience with PRI slips on Ericsson switches ? |
| 08:22.06 | Rawplayer | well the cpu is a p3 500 |
| 08:22.13 | Qwell | bit small |
| 08:22.16 | Rawplayer | hmm ok |
| 08:22.18 | Qwell | but it could work |
| 08:22.18 | IronHelix | esp if you do transcoding |
| 08:22.30 | Rawplayer | or a p2 dual 400mhz |
| 08:22.43 | Qwell | slightly better... |
| 08:23.05 | Qwell | without transcoding, you might be alright |
| 08:23.40 | Qwell | iax trunking might help with the vpn side of things |
| 08:23.47 | Qwell | not sure though |
| 08:24.16 | Qwell | though, all the calls are probably going to different places |
| 08:24.24 | Rawplayer | yes |
| 08:24.34 | Rawplayer | it will be like |
| 08:24.49 | IronHelix | client - vpn - * - provider |
| 08:24.53 | Rawplayer | student -----> asterisk/vpn server ------> other student of the project |
| 08:25.02 | Rawplayer | no not a provider |
| 08:25.07 | Qwell | Rawplayer: make sure you don't transcode, and you might be alright |
| 08:26.30 | Rawplayer | ok |
| 08:26.38 | Rawplayer | i need to read the asterisk book first |
| 08:26.44 | Qwell | rule of thumb: try it out. if it doesn't work, beef it up |
| 08:26.52 | Rawplayer | hehe |
| 08:30.06 | Qwell | off to bed |
| 08:30.22 | IronHelix | nite |
| 08:31.09 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 08:31.12 | zoa | hey ho ha |
| 08:31.21 | IronHelix | hi |
| 08:31.25 | Qwell | zoa: hi |
| 08:31.26 | Qwell | zoa: bye |
| 08:35.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:36.15 | zoa | ah yes i forgot the hi! |
| 08:38.44 | lme | bed..... |
| 08:45.38 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 08:47.07 | ikarus | anyone here have a suggestion for a simple Linux softphone for some testing |
| 08:49.00 | zoa | i could send you idefisk4linux if you want :) |
| 08:50.51 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:51.35 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 08:57.06 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 08:57.16 | zobia | Hello jollyr are there? |
| 08:57.47 | zobia | Hello everyone |
| 08:58.09 | zobia | anyone knows how to config incoming call screeing in the dialplan? |
| 08:58.44 | zobia | Incoming Call Screening. |
| 09:00.23 | infinity1 | if the database returns a string with a space, asterisk gives a WARNING when processing this line of code. any ideas of a workaround? for example, if the DB query sets the cidname to a string with a space |
| 09:00.28 | infinity1 | <PROTECTED> |
| 09:01.06 | infinity1 | i would classify this as a bug. should i report it? |
| 09:03.08 | infinity1 | if the DB stores a string with a "-" it chokes too!! omg. this is annoying. |
| 09:05.00 | *** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 09:06.41 | *** join/#asterisk Husk (n=Husk__@ppp115-183.static.internode.on.net) |
| 09:06.55 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 09:12.01 | *** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net) |
| 09:14.14 | ikarus | hrm, I have a silly problem, sometimes while doing a Playback or Echo and I hang up, the Asterisk server calls my SIP phone, is there anyway to prevent that ? |
| 09:14.17 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 09:14.49 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
| 09:18.05 | zoa | hey olle!!! |
| 09:25.18 | lme | infinity1: why don't you use the lookupcidname ? |
| 09:26.53 | *** join/#asterisk atif_ (n=atif@202.163.66.8) |
| 09:29.49 | infinity1 | lme: because it forces an exact match on CID |
| 09:29.59 | infinity1 | i don't want the +1 |
| 09:30.32 | lme | ok |
| 09:30.58 | infinity1 | so.. it turns out these are not equal (ignore the //) |
| 09:30.59 | infinity1 | <PROTECTED> |
| 09:30.59 | infinity1 | <PROTECTED> |
| 09:31.01 | lme | maybe you should try to modify the CALLERIDNUM before LookupCIDName |
| 09:31.46 | tzafrir_laptop | ikarus, iaxcomm is quite nice for testing |
| 09:32.15 | ikarus | tzafrir_laptop: I already have an app, but still the weird "call back" bug |
| 09:32.19 | tzafrir_laptop | e.g: convinient setup for multiple servers. Not the best of interfaces, though |
| 09:33.23 | tzafrir_laptop | I heard people recommend kiax, but haven't tried it myself. What call-back bug? |
| 09:34.38 | ikarus | tzafrir_laptop: I call an extension that hooks me up to Echo or Playback applications and when I hang up Asterisk calls my SIP phone and continues the application |
| 09:34.47 | ikarus | it happens at unpredictable moments |
| 09:34.51 | lme | infinity1: that's what i do to add or del some digits on my incoming numbers : Set(CALLERID(number)=00${CALLERIDNUM}) |
| 09:35.36 | tzafrir_laptop | ikarus, what sip phone? |
| 09:36.18 | ikarus | tzafrir_laptop: BudgeTone atleast (haven't got any other extensive phone to test, the softphone I now use lacks a numpad) |
| 09:38.26 | tzafrir_laptop | reminds me of a flash instead of a hangup |
| 09:39.07 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 09:39.21 | lme | infinity1: do you want an extract from my extensions.conf ? I use agi instead of lookupcidname, but callerid number modify is the same way... |
| 09:39.29 | ikarus | tzafrir_laptop: anyway to get asterisk to log exactly what it is doing ? |
| 09:40.10 | *** join/#asterisk gennaro (n=Paolo@ppp-62-10-136-43.dialup.tiscali.it) |
| 09:40.45 | gennaro | hi someone can explane me a think about compile kernel?!? |
| 09:41.01 | gennaro | i need to intall an x100p |
| 09:41.27 | gennaro | i istalled every rpm that i found as required |
| 09:41.40 | gennaro | and i'm donwloading my same scr kernel |
| 09:42.37 | gennaro | some one can help me? |
| 09:42.57 | infinity1 | lme: i did what you said. works :) |
| 09:43.13 | infinity1 | lme: I would have reported the bug, but it was taking too long. |
| 09:43.18 | lme | infinity1: great ! |
| 09:43.18 | InfraRed | gennaro: why not just use asterisk@home? |
| 09:43.21 | infinity1 | and i'm fucking tired. |
| 09:43.36 | infinity1 | lme: i still have to try agi. never got around to it. |
| 09:43.45 | gennaro | my boss don't like |
| 09:43.50 | gennaro | want asterisk |
| 09:43.53 | gennaro | .. |
| 09:43.57 | gennaro | too easy |
| 09:44.04 | lme | infinity1: i use it to look into an ldap structure (windows Active directory) for internal numbers |
| 09:44.04 | gennaro | :( |
| 09:44.14 | infinity1 | lme: thats a great idea |
| 09:45.34 | infinity1 | nite |
| 09:45.42 | lme | nite ! |
| 09:45.46 | lme | lucky one |
| 09:45.47 | InfraRed | Tue Nov 15 09:45:47 GMT 2005 |
| 09:46.00 | InfraRed | :) |
| 09:46.08 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 09:46.10 | lme | bouhouhouuuu |
| 09:46.20 | lme | begining my working day.... |
| 09:46.49 | *** join/#asterisk shog (n=rene@62.8.240.129) |
| 09:47.01 | shog | Hello everybody |
| 09:47.01 | InfraRed | laters |
| 09:47.29 | infinity1 | lme: er ..before i goto bed... one last thought. |
| 09:47.43 | shog | is there a way to change asterisk's behaviour regarding error codes? |
| 09:48.13 | infinity1 | we have a crm app that uses mysql. do you know where i can find an example agi script for cid lookup against it? preferably perl? |
| 09:48.44 | lme | infinity1: no sorry... I use perl scripts too, but only for ldap... |
| 09:49.19 | infinity1 | lme: i don't suppose i could use your script and replace the ldap sections with mysql |
| 09:49.22 | infinity1 | :) |
| 09:50.00 | lme | infinity1: why not ? just have to replace the ldap search into an sql one... But this is about 90% of the script :) |
| 09:50.43 | lme | infinity1: just look for Asterisk::AGI perl module |
| 09:51.00 | infinity1 | k |
| 09:51.33 | lme | infinity1: it's the easier way i found to interface perl with asterisk... |
| 09:51.48 | lme | consider i'm lazy as a pig |
| 09:54.01 | shog | i have an application, which i cannot change, that doesn't handle 503 errors correctly. So i need to change asterisk to send out 480 errors instead. Is this possible? |
| 09:54.24 | lme | imho you have to change it in the source code... |
| 09:55.30 | shog | lme: where should i look for the error handling? |
| 09:57.04 | lme | when you said 503 error code, you talked about sip correct ? |
| 09:58.34 | shog | yes |
| 09:59.12 | shog | i get a 503 error, when the line is busy |
| 09:59.14 | lme | shog : in your asterisk source tree. directory channel, file chan_sip.c |
| 09:59.18 | fourcheeze | is asterisk a "broken registrar" as far as Snom phones are concerned? |
| 09:59.48 | lme | shog : but you have several lines to modify |
| 10:01.21 | lme | shog : but it's very very crappy to modify this code... this is in accordance with rfc... and all update will erase your file if you use the head branch |
| 10:01.22 | shog | lme: can you tell me which lines need to be modified? |
| 10:02.38 | lme | shog : ouch.... I suggest you vi with /503 command... It's safer than me |
| 10:04.24 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 10:05.35 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 10:05.38 | lme | every times i touch a line in a c file, i fill my /tmp directory with core dump... That's like smoking.. I bet i better stop.... |
| 10:06.05 | Zeeek | no, just modify whole functions, never just 1 line :) |
| 10:06.27 | shog | i need to replace all occurences of 503 ? |
| 10:06.42 | lme | shog: yep... |
| 10:06.49 | shog | lme: thank you |
| 10:07.26 | lme | shog: <disclaimer> I DO NOT warranty that your * will not burn in a few minutes</disclaimer> |
| 10:07.44 | lme | but this should be tried |
| 10:07.49 | rLg | lol @ disclaimer |
| 10:09.54 | gennaro | what i must canghe to use an x100p? |
| 10:10.17 | lme | gennaro: ?? x100p in analogique fxo card correct ? |
| 10:10.35 | gennaro | ok an intel modem 56K |
| 10:10.44 | gennaro | with chipset supported |
| 10:10.46 | lme | argh |
| 10:11.01 | lme | i think that the intel chip is the only which is not supported |
| 10:11.11 | lme | ah ok |
| 10:11.37 | gennaro | i see on web that is a clone of x100p |
| 10:11.42 | lme | so you have not to change anything... zaptel support this card |
| 10:12.09 | gennaro | ah.. |
| 10:12.13 | lme | gennaro: yes... but there is a special chipset version at intel which is not supported... don't remember which ones. |
| 10:12.51 | gennaro | so i need to be able to have kernel source |
| 10:12.59 | lme | shog: what is your sip peer ? |
| 10:12.59 | gennaro | .config |
| 10:13.03 | lme | peers |
| 10:13.20 | gennaro | and anything else.. |
| 10:13.22 | gennaro | ??? |
| 10:13.44 | lme | why kernel sources ? what is your distro ? |
| 10:13.59 | gennaro | fedora c3 |
| 10:14.08 | gennaro | i see on voip-info |
| 10:14.14 | gennaro | some guides |
| 10:14.18 | gennaro | to install with fedora3 |
| 10:14.44 | gennaro | i'm tring... |
| 10:14.46 | lme | i don't think you have to recompile kernel.. |
| 10:14.49 | gennaro | i installed fc3 |
| 10:14.56 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 10:14.57 | lme | the only thing you'll need is pci support... |
| 10:15.01 | gennaro | i tried to compile asterisk |
| 10:15.19 | gennaro | and during that fase give error... |
| 10:15.36 | lme | which ? |
| 10:15.40 | gennaro | someone sayd try to follow the guide on voip... |
| 10:15.43 | rLg | gennaro: you'll need kernel source too |
| 10:16.14 | lme | yeah... in order to compile asterisk, you need kernel sources on your drive... |
| 10:16.36 | gennaro | i'm downloading the source code of my kernel from fc site |
| 10:16.47 | gennaro | on guide.. |
| 10:16.48 | lme | in fact, you need to have kernel sources on your drive to compile almost everything which is not 100% userland |
| 10:16.55 | gennaro | i need kernelsurce |
| 10:17.01 | gennaro | a copy of my config. |
| 10:17.32 | gennaro | so.. will be created a file required to compile zaptel |
| 10:18.13 | gennaro | is it true? |
| 10:18.19 | lme | there must be a package with kernel's sources of your distro |
| 10:19.22 | lme | gennaro: basically yes.... |
| 10:19.44 | gennaro | ok i'm tring.... |
| 10:19.55 | gennaro | i'm @34% of download ... |
| 10:20.06 | gennaro | i'm surfing @56 |
| 10:20.08 | rLg | also after installing kernel source.. you need to make a symlink /usr/src/linux-2.6 -> /lib/modules/`uname -r`/build |
| 10:20.16 | rLg | before compiling zaptel |
| 10:20.33 | gennaro | tanx |
| 10:20.43 | *** join/#asterisk langals (n=icechat5@196.7.14.183) |
| 10:23.09 | langals | Hi there....I am trying to install Asterisk 1.2.0-rc2 on Ubuntu linux. When I "make install" I get loads of errors like this: "chan_zap.c:10927: error: dereferencing pointer to incomplete type"...does anyone have any idea what could be causing this? |
| 10:23.14 | *** join/#asterisk Vhata (i=[U2FsdGV@platform.adept.co.za) |
| 10:25.03 | Vhata | what timezone does #asterisk live in? |
| 10:27.04 | zoa | 24/24 |
| 10:27.16 | Zeeek | hey zoa nice site! |
| 10:29.13 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 10:31.23 | lme | time to work with cdr ! |
| 10:32.39 | Vhata | I really want the new featureset in 1.2, but I'm a bit scared of the instability (I'm about to begin a project to replace our company's proprietary PBX). Any advice? |
| 10:32.48 | zoa | which one zeeek ? :) |
| 10:33.11 | Zeeek | ALl of 'em! No, the listing of voip providers. In fact I sent youa small correction |
| 10:33.28 | Zeeek | via "spamsucks2005" |
| 10:35.19 | zoa | ah yeah! |
| 10:35.27 | zoa | it should be corrected by now |
| 10:35.33 | zoa | i forwarded the email to the right person |
| 10:35.37 | Zeeek | what? That was 10 minutes ago! :) |
| 10:35.41 | zoa | working hard to make it even better |
| 10:36.05 | zoa | currently 5 people working on the database full time (althoug next week it will be 3 again) |
| 10:36.10 | Zeeek | I announced it on our private mailing list and a few people will undoubtedly contact you |
| 10:36.34 | Zeeek | voIP has really taken off here now that DSL is dirt cheap |
| 10:36.56 | Zeeek | OTH, every provider has a cheap voIP offering as well as TV via the same line |
| 10:37.01 | zoa | cool, thanks |
| 10:37.05 | Zeeek | np |
| 10:37.10 | zoa | i will try to add all destination prices too |
| 10:37.37 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 10:37.53 | *** join/#asterisk ful|work (n=fulgas@209.8.233.166) |
| 10:38.05 | cjk | hi, does anyone know if enumlookups are working in rc2. they were not working in rc1 an the betas |
| 10:40.52 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:43.12 | langals | anyone have any idea about that error I posted above? |
| 10:45.00 | zoa | zeeek, did you see the mailinglist archive we made too ? |
| 10:45.19 | Zeeek | I seem to remember something about it? What's the URL? |
| 10:45.26 | zoa | http://www.asteriskguru.com/archives/ -> its only 10 minutes delayed or so |
| 10:46.10 | Zeeek | nice! I'm actually reading one of the lists now in gmail |
| 10:46.12 | zoa | well, maximum 10 minutes delayed, we check our email account every 10 minutes i think |
| 10:46.29 | Zeeek | delivery is sometimes spotty though on the list itself |
| 10:46.30 | zoa | its similar to that, just with the complete archive there |
| 10:46.46 | zoa | most people didnt subscribe on day one :) |
| 10:47.54 | Zeeek | How true. If it's searchable it's very usable! |
| 10:48.57 | Zeeek | reading the huge Allison/Digium open source thread |
| 10:49.30 | zoa | it is seachable |
| 10:49.34 | zoa | searchable |
| 10:49.38 | Zeeek | I know |
| 10:49.46 | zoa | well, its not hyper fast yet, but working on that |
| 10:50.30 | Zeeek | I notied an interesting thing on yahoo ML: the search does about 10% of the messages and shows you results, asking if you need to go further |
| 10:50.44 | Zeeek | I found that a clever bit of thinking |
| 10:50.55 | ful|work | hey |
| 10:51.04 | Zeeek | I assume it looks first at the most recent, but I'm not sure |
| 10:53.44 | Zeeek | er, zoa? |
| 10:55.00 | Zeeek | is it possible thata search for 'polycom' didn't have any results in users? |
| 10:57.21 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 10:57.37 | zoa | no that doesnt sound ok |
| 10:57.39 | zoa | let me check that |
| 10:58.03 | zoa | earch found 1210 matches |
| 10:58.12 | zoa | seems to work for me |
| 10:58.19 | Zeeek | maybe it's a google cookie in my browser? |
| 10:58.36 | Zeeek | all I got was google ads |
| 10:58.38 | zoa | even the search polycon has resuls |
| 10:58.40 | zoa | aaah |
| 10:58.40 | zoa | no |
| 10:58.44 | zoa | use the built in search |
| 10:58.47 | zoa | not the google search |
| 10:58.52 | zoa | google is so slow with indexing |
| 10:58.54 | Zeeek | I was but isn't it google powered? |
| 10:58.59 | Zeeek | ooops I see. |
| 10:59.00 | zoa | http://www.asteriskguru.com/archives/search.php |
| 10:59.17 | Zeeek | the left sidebar search is google but that isn't the main search |
| 10:59.21 | zoa | yeah |
| 10:59.24 | zoa | thats just for the page |
| 10:59.29 | Zeeek | but you need it for revenue maybe? Otherwise I'd dump it |
| 10:59.30 | zoa | i will remove that one as soon as i can |
| 10:59.42 | zoa | i will dump it when i have a search for the tutorials |
| 10:59.46 | Zeeek | yeah it confused even an oldtimer like me |
| 10:59.48 | zoa | but for now its the best i can do |
| 10:59.51 | zoa | yeah i can imagine |
| 11:00.10 | Zeeek | that google logo really means "SEARCH" these days |
| 11:00.21 | zoa | revenue of google search is like 0,000000001$ a day or so :) |
| 11:00.37 | Zeeek | so you're onbviously not into porn |
| 11:00.41 | zoa | yeah |
| 11:00.50 | zoa | well, just imagine |
| 11:00.58 | zoa | if 1% of all people use that search |
| 11:01.07 | zoa | and of that 1% another 1% click on a banner |
| 11:01.14 | Zeeek | even my amazon stuff onmly makes enuff for a free book every 6 months :) |
| 11:01.35 | zoa | thats 1 for every 10.000 people visiting the site |
| 11:01.57 | zoa | so 20 cents or so for every 10.000 people |
| 11:02.14 | zoa | 10.000 people would be around 5 gb of traffic a day |
| 11:02.18 | Zeeek | actually, the wiki may get enough traffic tomake that worthwhile - but let's keepthat quiet |
| 11:02.36 | zoa | i dont think it could be |
| 11:02.57 | Zeeek | I know someone who has a site that makes a few bucks on those |
| 11:03.04 | zoa | on the search ? |
| 11:03.04 | Zeeek | but it has a huge userbase |
| 11:03.06 | zoa | weird |
| 11:03.11 | Zeeek | no, sorry on the ads |
| 11:03.24 | Zeeek | true it's not the same thing |
| 11:03.27 | zoa | yeah |
| 11:03.30 | zoa | ads are better |
| 11:03.46 | zoa | if its not for hurrycane katrina |
| 11:03.58 | Zeeek | because they are literally ubiquitous now, they piss me off too - even though they're easy to ignore |
| 11:04.04 | Dr_Ray | let jesus save your soul and win a freee ipod |
| 11:04.13 | Zeeek | Where do we sign? |
| 11:04.35 | *** join/#asterisk mazzanet_ (n=irc@unaffiliated/mazzanet) |
| 11:04.38 | zoa | yeah i know |
| 11:04.44 | zoa | me too |
| 11:04.44 | orlock | hey mazza |
| 11:04.49 | orlock | its safe here :) |
| 11:06.41 | *** join/#asterisk ciberbetics (n=bharatsa@210.211.246.47) |
| 11:06.49 | ciberbetics | hi everybody |
| 11:07.16 | ciberbetics | I am just configuring the Queues in Asterisk using real time... |
| 11:07.56 | ciberbetics | I have the table ready, but wondering do I need to specify some parameter in the queues.conf file or not... |
| 11:08.13 | ciberbetics | other than the configuration in the extconfig .conf file |
| 11:08.37 | ciberbetics | has anybody worked on the Asterisk Queues realtime |
| 11:08.42 | ciberbetics | please get back to me |
| 11:08.56 | *** join/#asterisk saftsack (n=saftsack@p54A7C949.dip.t-dialin.net) |
| 11:12.18 | *** join/#asterisk b000m (n=boom@opencode.tea.bg) |
| 11:13.09 | saftsack | are there any basetutorials for a simple asterisk installation? |
| 11:13.22 | Vhata | astlinux.org ? voip-info.org ? |
| 11:13.40 | saftsack | danke :) |
| 11:13.42 | saftsack | thank you |
| 11:14.53 | saftsack | voip-info.org is a huge website |
| 11:14.58 | saftsack | do you know a good articel? |
| 11:15.38 | Vhata | I'm gonna go with http://www.voip-info.org/wiki/index.php?page=Asterisk+installation+tips |
| 11:15.56 | b000m | can someone help me with configuring pick up groups? |
| 11:16.17 | saftsack | Vhata, thanks a lot :) |
| 11:17.05 | Vhata | saftsack: googling for 'asterisk installation tutorial' points you directly to http://www.asteriskguru.com/tutorials/ |
| 11:17.14 | Vhata | you *did* google before asking here, right? |
| 11:17.17 | saftsack | is there a debug mode for asterisk? i have a telephone yet but not a outgoing way to the internet |
| 11:17.29 | saftsack | can i test if the communication with the telephone works? |
| 11:18.41 | luite | just make an extensions with a Playback or something |
| 11:18.53 | luite | and enable a lot of debugging to the console in logger.conf |
| 11:19.09 | saftsack | ok :) |
| 11:19.26 | saftsack | and on the telephine itself there must stand the ip of the asterix server in dns and router? |
| 11:19.36 | b000m | does anybody know something about "Nothing to pick up" problem with pickupgroup? |
| 11:20.22 | luite | saftsack: you have to register to the asterisk server, you will see a message in the asterisk console, and with sip show peers you can check which phones have registered, and their ip's |
| 11:21.09 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:21.38 | X-Files | Hello ALL!!! I have big problem, I use eusso utg7104-22 and asterisk, in eusso all profiles (set coding 0 vad off) turn is off , but i call to asterisk and see errors : http://pastebin.com/430172 , Please help ! |
| 11:22.49 | saftsack | luite, ok i started my asterisk server on 10.10.10.16 |
| 11:23.26 | *** join/#asterisk NoRemorse (n=bah@202.161.68.2) |
| 11:23.34 | NoRemorse | ~seen justinnnn |
| 11:23.38 | jbot | justinnnn <~dsf@solid.mpa.net.au> was last seen on IRC in channel #asterisk, 238d 4h 6m 50s ago, saying: 'anyone ???'. |
| 11:23.49 | orlock | hey |
| 11:23.51 | NoRemorse | ~seen justinnnnn |
| 11:23.52 | jbot | justinnnnn <n=justinnn@61.95.68.85> was last seen on IRC in channel #asterisk, 35d 5h 58m 42s ago, saying: 'hey ppls :)'. |
| 11:23.54 | NoRemorse | heya orlock |
| 11:24.02 | orlock | what you up to these days? |
| 11:24.11 | NoRemorse | relaxing :) |
| 11:24.14 | orlock | chilling, or got plans? |
| 11:24.15 | orlock | ahh |
| 11:24.17 | orlock | cool :) |
| 11:24.47 | NoRemorse | playing with some voip stuff but mostly chiiling for awhile. |
| 11:24.57 | orlock | cool |
| 11:25.04 | orlock | you know NXT? |
| 11:25.15 | NoRemorse | nah? |
| 11:25.24 | NoRemorse | as in the dsl? |
| 11:25.29 | orlock | yup |
| 11:25.42 | mut | MMMmmm |
| 11:25.46 | mut | chicken noodle soup |
| 11:26.01 | NoRemorse | "at least I have chicken" |
| 11:27.48 | b000m | can someone help me with pick up groups and sip ? |
| 11:28.39 | saftsack | also i started asterix and i configured an ip to my sip telephone |
| 11:28.42 | saftsack | but nothing happens |
| 11:29.08 | many | it asterisk |
| 11:29.09 | NoRemorse | lol how more generic can u get |
| 11:29.10 | many | not asterix. |
| 11:30.00 | X-Files | Please help http://pastebin.com/430172 |
| 11:30.12 | saftsack | asterisk yes i know |
| 11:31.01 | luite | saftsack: has the phone registered with asterisk? |
| 11:31.20 | saftsack | no |
| 11:31.28 | saftsack | no output on the asterisk console |
| 11:31.38 | NoRemorse | saftsack try sip show peers |
| 11:31.50 | saftsack | nothing |
| 11:31.53 | luite | you have configured an account for the phone in sip.conf, right? then try sip show peers to check |
| 11:32.08 | saftsack | *CLI> -- Registered to '66.234.228.170', who sees us as 84.167.201.73:17838 |
| 11:32.08 | saftsack | sip show peers |
| 11:32.08 | saftsack | Name/username Host Dyn Nat ACL Mask Port Status |
| 11:32.13 | saftsack | that was the last |
| 11:33.44 | saftsack | do i have to enable this phone in my sip.conf? |
| 11:34.02 | NoRemorse | if you want security then yes |
| 11:34.15 | saftsack | this is a testing run ^^ |
| 11:34.28 | saftsack | how can i disable any security features? |
| 11:34.39 | saftsack | does one of you have a grandstream telephone? |
| 11:34.42 | NoRemorse | have a [guest] configd in sip.conf |
| 11:34.56 | NoRemorse | yes they are very easy to get working |
| 11:35.13 | luite | i only have snom and sipura, so can't help you with that |
| 11:35.32 | saftsack | howto type ip adresse with just 2 digits beetween the points into the telephone? |
| 11:35.51 | ikarus | saftsack: prefix with a 0 |
| 11:35.52 | Vhata | 012 ? |
| 11:36.12 | ikarus | 192168001042 |
| 11:36.14 | saftsack | yes i did it |
| 11:36.25 | saftsack | ok then its configured the right way |
| 11:36.43 | saftsack | i did 010.010.010.120 as the telephone adress |
| 11:36.57 | saftsack | 010.010.010.016 as the server adress (my computer with asterisk) |
| 11:37.16 | saftsack | and howto enable a guest account on my asterisk server now? |
| 11:38.00 | NoRemorse | sample sip.conf has it |
| 11:38.32 | saftsack | i havent got a sample sip.conf because i modified it so hard |
| 11:38.45 | NoRemorse | see /usr/src/asterisk |
| 11:38.56 | saftsack | ok |
| 11:39.02 | saftsack | now i have a sip.conf |
| 11:39.16 | saftsack | there is a general section |
| 11:39.48 | NoRemorse | no [guest] section |
| 11:40.08 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 11:40.15 | saftsack | no |
| 11:40.16 | X-Files | Nov 15 13:32:05 WARNING[7992]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? |
| 11:40.19 | X-Files | where problem ? |
| 11:40.21 | NoRemorse | argh ban |
| 11:40.40 | ikarus | NoRemorse: breathe |
| 11:41.10 | NoRemorse | just 1 line would have done just as well x |
| 11:41.12 | Rawplayer | re |
| 11:42.23 | X-Files | oh sorry :) this past from log ;/ next time |
| 11:44.52 | b000m | can someone help me with this : "NOTICE[23527]: chan_sip.c:10383 handle_request_invite: Nothing to pick up" ? |
| 11:45.05 | b000m | is there any guru? |
| 11:45.19 | NoRemorse | saftsack: see http://pastebin.ca/28777 |
| 11:45.41 | Zeeek | b000m you are dialing *8 ? |
| 11:45.42 | saftsack | thank you |
| 11:45.54 | b000m | Zeeek yes |
| 11:46.06 | Zeeek | and a call is ringing somewhere else? |
| 11:46.19 | b000m | yes , there is a call |
| 11:46.25 | NoRemorse | anyone know of a 2line dect cordless phone? |
| 11:46.32 | b000m | but i can't pick it up |
| 11:46.38 | Zeeek | b000m what are the phones? |
| 11:46.47 | saftsack | so im in school now |
| 11:46.51 | saftsack | Cya and thankyou |
| 11:46.52 | b000m | everithing is fine in sip.conf |
| 11:47.13 | b000m | cisco 7905G and some soft phones too |
| 11:47.33 | b000m | they are all in pickupgroup=1 |
| 11:49.26 | b000m | soft phones are x-lite |
| 11:49.49 | b000m | but i think that it is not from the pfones |
| 11:50.02 | Zeeek | what document did you find pickupgroup in? |
| 11:51.49 | b000m | Zeeek, http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 11:53.39 | NoRemorse | omgwtfboringbbq |
| 11:54.31 | Zeeek | b000m and what is the callgroup of the phones? |
| 11:55.11 | b000m | no callgroup |
| 11:55.19 | b000m | just pickupgroup |
| 11:55.41 | ikarus | hmmm, anyone managed to get more then 3 numbers with a distinct ringtone into a BudgeTone, without faking the ID |
| 11:55.59 | Zeeek | well according to the page you just showed.... |
| 11:56.00 | Zeeek | <PROTECTED> |
| 11:58.08 | b000m | Zeeek, you are right |
| 11:58.36 | b000m | I will try with callgroup=1 at every entry in my sip.conf |
| 11:59.17 | b000m | may be that is the clue |
| 12:03.48 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:03.57 | RoyK | 10 PRINT "HELLO WORLD" |
| 12:05.42 | luite | 20 GOTO 10 |
| 12:05.42 | luite | that was my first program, from the C64 manual |
| 12:06.44 | Vhata | did it work? |
| 12:07.09 | luite | yup, but I had to press the break key to stop it :) |
| 12:07.23 | Vhata | you and your fancy "break key" technology |
| 12:12.56 | *** join/#asterisk coppice (n=chatzill@7.206.17.210.dyn.pacific.net.hk) |
| 12:12.59 | *** join/#asterisk juanjoc (n=juanjoc@OL48-53.fibertel.com.ar) |
| 12:21.39 | mut | hmmm |
| 12:21.43 | mut | is this possible |
| 12:22.17 | mut | i need to make an online application process for customers for 5 different types of services; dialup, wireless, dsl, phone service, voip |
| 12:22.29 | mut | think i could somehow make that into one application? |
| 12:23.15 | mut | and theres always the little things with each one, web accelerators, security suite, call features for regular phone service and voip |
| 12:23.21 | mut | extra emails |
| 12:23.26 | mut | umm |
| 12:24.24 | *** join/#asterisk stevie20 (n=stevie@mini.fdknet.de) |
| 12:25.17 | stevie20 | hello |
| 12:25.27 | stevie20 | lme ? |
| 12:28.04 | *** join/#asterisk tengulre (n=tengulre@61.150.12.86) |
| 12:28.49 | juanjoc | Hi, has someone used RxFAX and TxFAX (as caller) to talk to each other? They don't seem to cope with each other very well... |
| 12:28.51 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:30.53 | coppice | juanjoc: over a T1 or E1 circuit they will talk. connected directly together inside *, limitations in * mean they do not |
| 12:31.02 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 12:31.06 | lehel | hello |
| 12:31.35 | juanjoc | coppice: I'm using two instances of Asterisk connected via SIP |
| 12:31.58 | juanjoc | coppice: The problem is that both application block waiting for frames from the other one |
| 12:32.49 | coppice | FAX over IP does really work, so I don't even bother trying to make it |
| 12:33.52 | *** join/#asterisk gennaro (n=Paolo@ppp-62-10-136-89.dialup.tiscali.it) |
| 12:33.58 | juanjoc | I know, but the problem here is not related to the use of SIP |
| 12:34.10 | gennaro | hi |
| 12:34.12 | gennaro | i downloaded surce code of my kernel i hope.. |
| 12:34.13 | juanjoc | The apps don't transmit any frames |
| 12:34.19 | gennaro | and i tried to copi |
| 12:34.38 | coppice | juanjoc: you ignore the part after the comma |
| 12:34.45 | gennaro | config-.... in usr/src/.. /.config |
| 12:34.53 | gennaro | make menuconfig |
| 12:34.58 | gennaro | dont work |
| 12:35.02 | gennaro | why? |
| 12:36.18 | *** join/#asterisk Tili (n=Tili@83.110.223.215) |
| 12:37.56 | b000m | gennaro, why you do that? |
| 12:38.14 | juanjoc | coppice: Maybe I misunderstood, but I'm using RxFAX and TxFAX in different instances of Asterisk on different machines |
| 12:38.28 | gennaro | in order to compile zaptel |
| 12:38.40 | gennaro | my guide said so.. |
| 12:38.45 | gennaro | download kernel |
| 12:39.06 | coppice | to make them send at the right rate when connected to a PSTN trunk then send in sympathy with what they receive. if they receive nothing, they send nothing |
| 12:39.13 | gennaro | put it on usr/src/linux |
| 12:39.41 | gennaro | cp /boot/conifg-n.n.n as ../linux/.config |
| 12:39.45 | gennaro | so |
| 12:39.50 | gennaro | make menuconfig |
| 12:40.02 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
| 12:40.08 | gennaro | .. |
| 12:40.18 | gennaro | what should i do?!? |
| 12:40.44 | b000m | where did you copy it exactly and what is your distro? |
| 12:40.53 | juanjoc | coppice: But wouldn't one of the apps send the initial fax tones regardless of what the other one does? |
| 12:41.01 | gennaro | i'm using an fc3 |
| 12:41.19 | coppice | nope. they only send when they receive |
| 12:42.19 | gennaro | i downloaded my same kernel from fc |
| 12:42.51 | gennaro | rpm -Uvh kernel0.0.0.0.. .src.rpm |
| 12:43.11 | gennaro | i created usr src redhat |
| 12:43.18 | gennaro | and usr scr redhat sources |
| 12:43.54 | b000m | extract the sources in /usr/src |
| 12:44.27 | b000m | and ln -sf source_dir linux |
| 12:44.28 | gennaro | i extracted linux-2.6.9.tar.bz2 |
| 12:44.28 | gennaro | <PROTECTED> |
| 12:45.05 | gennaro | so i have an directory named linux-2.6.9 |
| 12:45.13 | b000m | ok |
| 12:45.18 | gennaro | where i have i hope my source kernell |
| 12:45.20 | b000m | do this |
| 12:45.26 | RoyK | gennaro: er... 2.6.9 is like 13 months old |
| 12:45.31 | gennaro | ok |
| 12:45.31 | gennaro | ... |
| 12:45.46 | gennaro | but i have only a cd on this machine.. |
| 12:45.50 | b000m | ln -sf linux-2.69 linux |
| 12:46.02 | gennaro | and i0m working with 56k |
| 12:46.12 | b000m | and then copy your config file there |
| 12:46.15 | gennaro | ? |
| 12:46.23 | gennaro | i'm going.. |
| 12:46.42 | RoyK | genmud: the patches aren't that large |
| 12:46.57 | RoyK | ftp://ftp.kernel.org/pub/linux/kernel/v2.6/patch-2.6.10.bz |
| 12:46.58 | RoyK | etc |
| 12:47.05 | RoyK | bz2, even |
| 12:49.16 | gennaro | i do so.. |
| 12:49.21 | gennaro | and nothing is changed |
| 12:49.35 | gennaro | b000 what can i do? |
| 12:51.33 | syle | don;t have all day to update kernels on a million machines just do yum update kernel |
| 12:51.40 | syle | something like that |
| 12:52.18 | gennaro | what?!? |
| 12:52.28 | gennaro | tried to do yum ... |
| 12:52.44 | gennaro | but.. i'll stay a week.. @56k |
| 12:53.38 | gennaro | and if i lunch yum kernelsource ? |
| 12:55.59 | syle | yum install kernel-dev |
| 12:56.00 | gennaro | in other kind of way |
| 12:56.10 | syle | or source |
| 12:56.13 | syle | check google |
| 12:56.47 | gennaro | in other kind of way rpmbuild --target i686 ... |
| 12:57.11 | gennaro | but i havent an rpmbuild and doesn't work as rpm build |
| 12:57.38 | gennaro | i'm oing to try with yum.. |
| 13:05.27 | cjk | is it possible to bind asterisk to more than one port? |
| 13:07.52 | lme | damn |
| 13:08.09 | lme | can't we use ; delimiters for cdr_custom ? |
| 13:11.07 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:13.35 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 13:16.47 | RoyK | ~seen zoa |
| 13:16.50 | jbot | zoa is currently on #asterisk (4h 45m 41s). Has said a total of 54 messages. Is idling for 2h 12m 6s |
| 13:18.10 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 13:19.12 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 13:19.12 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:19.38 | *** join/#asterisk dfriend (i=dfriend@69.89.173.131) |
| 13:23.23 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 13:27.06 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 13:30.42 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-235-70.44-151.net24.it) |
| 13:40.16 | X-Files | heya all |
| 13:40.26 | BladeRunner05 | hi x |
| 13:40.28 | X-Files | Nov 15 15:33:01 DEBUG[9947]: chan_sip.c:7380 handle_request: Check for res for 75305101 |
| 13:40.29 | X-Files | Nov 15 15:33:01 DEBUG[9947]: chan_sip.c:1629 update_user_counter: Call from user '75305101' is 1 out of 0 |
| 13:40.35 | *** join/#asterisk Druken (i=Druken@67.69.139.226) |
| 13:40.41 | Druken | morning everyone |
| 13:40.46 | X-Files | extensions.conf : |
| 13:40.59 | X-Files | [default] |
| 13:41.01 | X-Files | exten => s,1,Answer() |
| 13:41.05 | Druken | ~pastebin |
| 13:41.09 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 13:41.09 | X-Files | why not answer ? |
| 13:41.19 | X-Files | ok |
| 13:41.22 | Druken | ahh.... hehe |
| 13:41.35 | Druken | what kinda device are ya using? |
| 13:41.42 | Druken | plus remove the () |
| 13:42.04 | BladeRunner05 | the latest chan_capi version is 0.3.5 ? |
| 13:42.24 | X-Files | http://pastebin.ca/28784 |
| 13:42.26 | X-Files | please check |
| 13:43.55 | X-Files | no comments ? |
| 13:44.00 | Druken | X-Files: so it's a sip call ? |
| 13:44.10 | lehel | BladeRunner05: http://www.junghanns.net/en/home.html |
| 13:44.21 | ikarus | x |
| 13:44.30 | luite | BladeRunner05: you can get the newer chan_capi-cm from sourceforge, the latest version is 0.6.1 |
| 13:44.43 | X-Files | Druken: yep this SIP |
| 13:44.50 | X-Files | ikarus: ? |
| 13:45.07 | ikarus | X-Files: oh, I just dropped a tray of parts on my keyboard |
| 13:45.17 | X-Files | ;) |
| 13:45.18 | Druken | X-Files: well, what are you expecting it to do ? so far your just answering the call... not doing anything after that.... |
| 13:46.42 | X-Files | Druken: not answer asterisk and droped connection (line busy) |
| 13:47.27 | *** join/#asterisk cmslaght (n=cmslaght@admin.ambt.net) |
| 13:48.35 | Druken | X-Files: what is it doing? |
| 13:49.13 | X-Files | Druken: check log : http://pastebin.ca/28784 |
| 13:49.27 | BladeRunner05 | lehel: and what about this version http://sourceforge.net/projects/chan-capi |
| 13:50.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:51.06 | Druken | i looked at the pastebin... but your only showing an answer, so asterisk will answer... and then basically hang it up.. cause it doesn't have anything left to do... the call will just sit there, |
| 13:51.51 | X-Files | ok for testing i add background sound |
| 13:51.53 | X-Files | wait |
| 13:52.36 | Druken | answer then background(invalid) |
| 13:52.37 | Druken | :) |
| 13:52.54 | X-Files | demo :) |
| 13:53.41 | lehel | BladeRunner05: i've heard chan_capi-cm-0.6 can't be loaded with latestasterisk version from cvs, but give it a try.. everything is possible, and tell me if you succeded |
| 13:54.26 | X-Files | try check : http://pastebin.ca/28785 |
| 13:54.46 | X-Files | i update extension file and restart asterisk |
| 13:54.55 | X-Files | result null ;( |
| 13:55.40 | Druken | set verbose 12 on cli |
| 13:55.50 | Druken | sip no debug |
| 13:55.53 | X-Files | ok |
| 13:55.57 | Druken | and then try your call |
| 13:56.05 | Druken | see what asterisk is actually doing |
| 13:56.37 | X-Files | this same |
| 13:57.11 | *** join/#asterisk Broom (i=Broom@jescobar.ayustar.net) |
| 13:57.16 | Druken | paste me the cli output |
| 13:57.22 | X-Files | wait |
| 13:57.57 | X-Files | http://pastebin.ca/28787 |
| 13:58.03 | X-Files | check |
| 13:59.17 | Druken | X-Files: sorry, can't help ya... |
| 13:59.23 | X-Files | ;( |
| 13:59.58 | Broom | hello, i have a terrible crackling noise (not always, just sometimes) on calls coming in through a digium Wildcard TE110P T1, what can I verify? |
| 14:00.38 | BladeRunner05 | lehel:ok, I tell u... |
| 14:00.38 | lme | Broom: what does a zap show status display about your te110 ? |
| 14:01.26 | Broom | lme: zap show status? or zap show channels? |
| 14:01.32 | lme | status |
| 14:01.42 | Broom | there is no option for status |
| 14:01.51 | lme | no |
| 14:01.54 | Broom | on the cli i mean |
| 14:01.54 | lme | zap show status |
| 14:01.58 | lme | as it |
| 14:02.06 | Broom | No such command 'zap show status' (type 'help' for help) |
| 14:02.27 | lme | ouch |
| 14:02.45 | lme | asterisk*CLI> zap show status |
| 14:02.45 | lme | Description Alarms IRQ bpviol CRC4 |
| 14:02.52 | lme | Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 |
| 14:02.52 | Broom | what ver/. |
| 14:03.06 | lme | CVS-D2005.05.28.22.00.00-11/14/05-09:45:09 |
| 14:03.11 | lme | bristuffed |
| 14:03.21 | lehel | ~pastebin lme |
| 14:03.27 | Broom | i dont have bristuff |
| 14:03.30 | Broom | installed |
| 14:03.48 | lme | lehel: ? |
| 14:04.06 | Broom | i have 1.0.9 |
| 14:04.14 | Broom | downloaded from webpage not cvs |
| 14:04.40 | *** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
| 14:04.41 | lme | okay |
| 14:04.42 | *** part/#asterisk shog (n=rene@62.8.240.129) |
| 14:04.59 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 14:05.07 | lme | Broom: just try to make zttool in your zaptel dir, launch it, and display info about your card |
| 14:05.33 | Broom | ok, wait |
| 14:06.27 | lme | lehel: (15:03:21) lehel: ~pastebin lme ?? |
| 14:06.54 | lehel | lme, just use pastebin |
| 14:07.04 | lehel | ~pastebin |
| 14:07.05 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 14:07.05 | Broom | lme: they dont like ppl who paste in the channel, pastebin is an utility for doing so |
| 14:07.28 | lehel | yeah Broom Broom.. |
| 14:07.33 | lme | okay okay |
| 14:07.36 | lme | sorry for that |
| 14:07.37 | *** join/#asterisk evangelion (n=manzy_ze@213.199.26.99) |
| 14:07.47 | evangelion | hi all |
| 14:08.11 | Broom | lme, what information from zttool you want me to give you? |
| 14:08.15 | Katty | hihi (= |
| 14:08.55 | lme | Broom: have you got irq misses ? |
| 14:09.15 | Broom | yes |
| 14:09.22 | lme | blam |
| 14:09.25 | Broom | 159371 |
| 14:09.29 | lme | ouch |
| 14:09.35 | Broom | yeah, any suggestions? |
| 14:09.58 | mut | get a new motherboard |
| 14:10.05 | Broom | jaja |
| 14:10.18 | Broom | brand/model? |
| 14:10.20 | mut | turn off everything in the bios |
| 14:10.27 | mut | thats using a irq |
| 14:10.33 | lme | Broom: lspci -vv and look for shared irq |
| 14:10.46 | evangelion | does any of you run asterisk realtime with "clustering" patch? |
| 14:10.52 | mut | usb/serial/audio/lan |
| 14:10.56 | Broom | ok, lemme do that |
| 14:11.12 | Broom | it is an intel board |
| 14:11.41 | lme | Broom: kernel hand made or from your distro ? |
| 14:11.53 | mut | just |
| 14:11.58 | mut | cat /proc/interrupts |
| 14:12.08 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 14:12.27 | mut | easier to tell |
| 14:12.34 | lme | for sure |
| 14:13.46 | Broom | ok, hold on |
| 14:14.13 | Broom | http://pastebin.ca/28791 |
| 14:14.16 | Broom | thats what i got |
| 14:14.26 | Broom | kernel is from distro |
| 14:14.37 | lme | bouhouhou |
| 14:14.48 | lme | takes me about 5 mins to load this page |
| 14:15.35 | lme | mmfmm |
| 14:15.45 | evangelion | does any of you run asterisk realtime with "clustering" patch? |
| 14:15.53 | lme | evangelion: no sorry |
| 14:16.17 | lme | Broom: okay... seems like your te110 card actually sharing irq with your usb root |
| 14:16.56 | Broom | ok, thanks i'm disabling the usb,audio etc as of now |
| 14:17.01 | lme | Broom: so let's start from the beginning, kill all unnecessary in your bios |
| 14:17.21 | ikarus | No APIC ? |
| 14:17.32 | lme | Broom: if you still get irq sharing (which might be magical thing) you should use IO-APIC in your kernel |
| 14:18.10 | X-Files | Druken: maybe problem in sip.conf ? |
| 14:19.05 | lme | ikarus: apparently, distro kernel... |
| 14:19.17 | cjk | hi, does anyone know if enumlookups are working in rc2. they were not working in rc1 an the betas |
| 14:19.25 | ikarus | lme: hmmmmm, I guess, I never use anything else |
| 14:21.30 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 14:21.46 | lme | ikarus: what distro are you using ? |
| 14:22.13 | ikarus | lme: Debian, but I don't use distro kernels on anything but the most basic box |
| 14:22.16 | X-Files | Help please , try connect to asterisk from phone sip, but asterisk not answer , see log : http://pastebin.ca/28787 |
| 14:22.32 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 14:22.47 | *** part/#asterisk juanjoc (n=juanjoc@OL48-53.fibertel.com.ar) |
| 14:22.58 | lme | ikarus: okay... we're agree :) |
| 14:23.05 | lme | debian too here |
| 14:23.32 | *** join/#asterisk juanjoc (n=jcomella@OL48-53.fibertel.com.ar) |
| 14:26.55 | *** join/#asterisk JimmyCarter (n=del@213083175015.sonofon.dk) |
| 14:27.25 | JimmyCarter | anyone know how to periodically reset the queues? |
| 14:27.40 | JimmyCarter | like every 15 minutes or so. |
| 14:28.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:32.48 | *** join/#asterisk dalabera (n=dalabera@146.82.190.164) |
| 14:33.48 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 14:36.20 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 14:38.19 | tmccrary | I have a problem with ivr on asterisk |
| 14:38.47 | tmccrary | My asterisk seems to be very poor at understanding dial tones for the menu |
| 14:39.22 | ikarus | tmccrary: inband with compression on ? |
| 14:40.28 | stevie20 | hi lme... |
| 14:40.29 | tmccrary | I dont have anything like that in my sip.conf or extensions.com |
| 14:40.46 | tmccrary | exten => s,1,DigitTimeout(10) |
| 14:40.46 | tmccrary | exten => s,2,Background(initial-menu) |
| 14:40.46 | tmccrary | exten => 9,1,Goto(ivr-main,s,2) |
| 14:40.55 | tmccrary | that's my dial plan, am I missing a required field or anything |
| 14:41.59 | tmccrary | like do I need to set a field that increase the time it waits for digits? that's what I set the digittimeout for, but it doesn't work well |
| 14:42.49 | lme | hi stevie20 ! |
| 14:42.49 | tmccrary | like if I press the numbers very fast, it works usually |
| 14:44.25 | JimmyCarter | anyone know how to periodically reset the data in the queues? like servicelevelperf etc. |
| 14:44.48 | lme | no sorry |
| 14:45.04 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 14:45.58 | evangelion | does any of you run asterisk realtime with "clustering" patch? |
| 14:48.03 | stevie20 | lme, do you have the possibility, to test the beta version with a patch? you remember, the silence supression problem? |
| 14:50.01 | stevie20 | and after upgrading to 1.0.9 i experienced a new problem... |
| 14:50.40 | stevie20 | we have got some calls, which are getting forwarded to another number... |
| 14:50.53 | stevie20 | but the forwarding should not be done wit asterisk... |
| 14:51.55 | stevie20 | a incoming call, wich gets forwarded looks like: |
| 14:53.01 | lme | stevie20: i'm bristuffed, i canno't follow the head branch |
| 14:53.29 | stevie20 | PSTN - incoming call to --> Asterisk -- forwarded to --> Old PBX -- new (incoming) Call --> Asterisk -- forwarded to --> SIP Gateway --> PSTN --> final destination |
| 14:53.52 | LostFrog | lol.. bristuffed.. sounds like a disease. |
| 14:53.59 | stevie20 | ok lme... |
| 14:54.50 | stevie20 | and now, the connection get lost at the first incoming call.... with the following error message: |
| 14:55.33 | stevie20 | Nov 15 15:46:19 WARNING[5720]: Unable to forward frame |
| 14:56.08 | lme | LostFrog: and so it is.... but here in france, for smi, we use to have many T0 incoming instead of pri |
| 14:56.10 | stevie20 | in the meanwhile the other calls are setup and the destination devic is ringing about 1 to 2 times... |
| 14:56.24 | lme | LostFrog: so i need octo + quad bri from junghanns |
| 14:57.03 | stevie20 | in 1.0.8 i got timeouts... i increased the timeout value and it works... but this hack doesnt seems to work with 1.0.9 |
| 14:57.38 | *** join/#asterisk Anthro (n=dsfgrt@pdpc/supporter/active/Anthro) |
| 14:57.44 | Anthro | Is anyone using BroadVoice? |
| 14:58.06 | lme | stevie20: damn, your incmoing calls looks like hardworker to get established ! |
| 14:59.01 | stevie20 | lme say it loud ;-) |
| 14:59.55 | stevie20 | but this is "only" a test scenario, to test, if asterisk will work with our setup... i hope the final customers don't need this setup... |
| 15:01.15 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 15:01.15 | *** mode/#asterisk [+o twisted] by ChanServ |
| 15:03.01 | *** join/#asterisk Broom (i=Broom@jescobar.ayustar.net) |
| 15:04.03 | *** join/#asterisk hugov6 (n=foo@p54AD5733.dip.t-dialin.net) |
| 15:04.06 | hugov6 | hiho |
| 15:04.40 | *** join/#asterisk ful|work (n=fulgas@209.8.233.238) |
| 15:05.06 | Broom | lme: i changed the bios settings here is my cat /proc/interrupts outcome |
| 15:05.19 | Broom | http://pastebin.ca/28798 |
| 15:06.56 | Broom | and there are no irq misses, but i still hear a crackling noise on some calls |
| 15:07.32 | hugov6 | does a if statement exist in extensions.conf? |
| 15:07.49 | hugov6 | or something? |
| 15:07.57 | LostFrog | hugo-v6: GotoIf? |
| 15:08.17 | LostFrog | hugo-v6: show applications |
| 15:08.37 | hugov6 | LostFrog hmmm maybe this would be possbile. ill have a look. thank you. |
| 15:09.07 | LostFrog | You might also want to look at Asterisk Variables on the wiki. |
| 15:09.15 | LostFrog | And Asterisk Expressions |
| 15:09.33 | hugov6 | thank you LostFrog. |
| 15:11.53 | *** join/#asterisk DYOGI_B (n=Jade@dsl-202-173-190-245.qld.westnet.com.au) |
| 15:12.12 | DYOGI_B | hey what is the best codec to use on Asterisk in a LAN environment |
| 15:12.56 | LostFrog | ulaw. |
| 15:13.21 | DYOGI_B | and is alaw just as good? |
| 15:13.25 | LostFrog | well.. ulaw or alaw.. I'm not in europe, so I don't know what the advantages of alaw would be. |
| 15:14.07 | DYOGI_B | thanks :) Australia i think it uses Alaw |
| 15:14.18 | hugov6 | LostFrog: gotoif will work. thank you. |
| 15:14.24 | DYOGI_B | ok i will go fix all the sip phone on my network now :) |
| 15:18.31 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 15:19.13 | tmccrary | Anthro |
| 15:19.16 | tmccrary | I am using BroadVoice |
| 15:19.37 | Anthro | tmccrary: Is your "device" set to Generic SIP device? |
| 15:20.08 | tmccrary | You mean in the broadvoice control panel thing? |
| 15:20.09 | DYOGI_B | What do people think of the grandstream 2000 |
| 15:20.10 | tmccrary | let me check |
| 15:20.23 | tmccrary | DYO: it's ok, nothing to write home about... cheap though |
| 15:20.44 | tmccrary | I assume you mean the GXP-2000, I have one sitting right here |
| 15:20.58 | Anthro | tmccrary: Also, it looks like the MAC address doesn't matter, but I'm not sure. |
| 15:21.13 | DYOGI_B | yep, i have bough 4 for an office install i have to do, do you think they will be ok ? |
| 15:21.28 | tmccrary | depends on how picky your customer is |
| 15:21.37 | [TK]D-Fender | Granstream = bleagh... |
| 15:21.39 | tmccrary | the speakerphone is attrocious |
| 15:22.08 | DYOGI_B | is there much echo as i am trying to fix it |
| 15:22.33 | tmccrary | Anthro: I have GENSIP-XXXXXXXXXXXXXX in my devices config |
| 15:22.48 | kuku5 | anyone using paging speakers ? |
| 15:22.50 | Anthro | tmccrary: Okay, thanks. |
| 15:23.13 | [TK]D-Fender | Flimsy feel, flakey transfer (IIRC) and bunch of other "downs". SPA-941 is more expensive but worth it. |
| 15:23.13 | tmccrary | it connects to my colo server in texas, works okay. Sometimes I've had new calls get dropped though, on BroadVoice's end |
| 15:23.36 | tmccrary | I'm not a big fan of sipura |
| 15:23.44 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 15:24.13 | tmccrary | Of their last two phones I've used, I hated them both and they had weird buzzing noises (that none of the other phones I tested had.. grandstream and snom) |
| 15:24.14 | DYOGI_B | hmm |
| 15:24.32 | yxa | do fxo ports require the 5V supply to be connected on the tdm400p? |
| 15:25.05 | [TK]D-Fender | tmccrary : SPA-841's I take it? |
| 15:25.55 | DYOGI_B | yep 5v |
| 15:26.05 | DYOGI_B | err no just the fxs |
| 15:26.23 | yxa | DYOGI_B i heard the new tdm2400p doesnt need the 5V for fxo :) |
| 15:26.39 | file | eh? |
| 15:26.42 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 15:26.49 | yxa | errr fxs |
| 15:26.56 | yxa | no i mean fxo |
| 15:26.58 | tmccrary | D-Fender: let me check, I have those phones in a closet :) |
| 15:27.12 | file | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P&tab=details notice the power connector? ;) |
| 15:27.46 | luite | but not required for fxo |
| 15:27.47 | DYOGI_B | didn't he say the tdm400p |
| 15:28.02 | DYOGI_B | fxo no fxs yes |
| 15:28.21 | Anthro | tmccrary: I think http://www.broadvoice.com/support_install_asterisk.html is wrong about the format of the register string. Shouldn't it be register => <phonenumber>:<password>@sip.broadvoice.com:<phonenumber>@sip.broadvoice.com/<extension>? |
| 15:30.06 | yxa | DYOGI_B thanks |
| 15:31.11 | DYOGI_B | w |
| 15:33.26 | *** join/#asterisk steff (n=steff@80.125.254.220) |
| 15:33.31 | steff | hi all |
| 15:34.33 | steff | somebody know a goo 2TO card for asterisk, i have some problems with an AVM C2 :-( |
| 15:35.38 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 15:36.21 | steff | anyone ? |
| 15:38.18 | *** join/#asterisk DCGrendel (n=DCGrende@ip68-1-157-197.mc.at.cox.net) |
| 15:38.33 | DCGrendel | anyone here run * under xen? |
| 15:40.26 | DCGrendel | i'm having trouble with everything that relies on a timer, even with ztdummy loaded. |
| 15:40.47 | *** join/#asterisk shout (n=rcsw@host213-123-195-3.in-addr.btopenworld.com) |
| 15:41.11 | DCGrendel | not to mention voipbuster refuses to route my calls from *, but works fine when using their app |
| 15:41.26 | [TK]D-Fender | 2TO? |
| 15:42.16 | steff | 4 isdn channels |
| 15:42.30 | steff | 2 bri |
| 15:43.59 | luite | don't know about any 'double' card, you should probably get a quad from junghanns or beronet |
| 15:50.49 | hugov6 | iirc was there a application(?) for extensions.conf which shows the content of a variable? |
| 15:50.50 | steff | AVM C2/C4 are the same, i can't get more than 2 chan vi chan_capi, and with misdn the card will not work, i miss something, but i can't find docs about this type of setup |
| 15:50.51 | hugov6 | ah noop |
| 15:50.53 | ManxPower | DCGrendel, Then you have some other problem. |
| 15:50.53 | DCGrendel | ManxPower: i dont pretend to know what the problem is, but it might have something to do with the HZ variable not being 1000 on my kernel. |
| 15:50.57 | ManxPower | DCGrendel, That would do it. |
| 15:50.57 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 15:52.55 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 15:54.30 | *** join/#asterisk graphyx (n=mike@67.50.46.118) |
| 15:54.42 | graphyx | anyone have any recommendations for linux softphone for SIP? |
| 15:55.13 | jhb | graphyx: I like to use sjphone |
| 15:55.27 | tmccrary | linphone works okay |
| 15:55.34 | jhb | not open, but works really reliable for me |
| 15:55.38 | tmccrary | I don't like softphones in general |
| 15:55.40 | }cytrak{ | why does asterisk use gsm format ? |
| 15:55.47 | tmccrary | you can use other codecs |
| 15:55.52 | }cytrak{ | isn't that computational intensive ? |
| 15:55.57 | tmccrary | gsm is just an old reliable standard that's open |
| 15:56.14 | }cytrak{ | what else can be used ? wav ? |
| 15:56.17 | tmccrary | it doesn't have the best quality, but it can get the job done and it's not encumbered by patents |
| 15:56.18 | graphyx | tmccrary: Any luck using linphonec ? I can't figure out how to use the console to configure it to work with asterisk. |
| 15:56.24 | tmccrary | pcmu/ulam |
| 15:56.33 | tmccrary | or (alaw if you're in europe) |
| 15:56.42 | tmccrary | wav is really a wrapper for codecs |
| 15:56.47 | }cytrak{ | i c |
| 15:56.48 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 15:56.53 | tmccrary | s/ulam/ulaw |
| 15:57.10 | tmccrary | gsm is low bit rate, low quality |
| 15:57.16 | }cytrak{ | i'm trying to convert my wav to gsm as the asterisk book mentioned by my sox doesn't support gsm |
| 15:57.16 | tmccrary | its lossy compression |
| 15:57.34 | Rav1974 | Hi guys, I have a silly question, which is the latest bleeding edge version? The CVS checkout or RC2? |
| 15:57.40 | graphyx | CVS |
| 15:57.43 | LostFrog | CVS |
| 15:57.45 | Vhata | always CVS |
| 15:57.46 | tmccrary | ulaw is lossless, so it takes up mega bandwith |
| 15:57.51 | LostFrog | The answer to that question is always CVS. |
| 15:57.51 | ManxPower | tmccrary, It's no worse than a good cellphone call. |
| 15:58.03 | Rav1974 | thanks guys |
| 15:58.12 | tmccrary | right, but its not as good as ulaw (which is what a regular land line phone will use) |
| 15:58.31 | ManxPower | tmccrary, Correct. But there are alternatives. |
| 15:58.43 | tmccrary | what codec you use is really dependant on how much bandwidth you want to use |
| 15:58.44 | DCGrendel | anyone have viopbuster in their dialplan? |
| 15:58.49 | Vhata | LostFrog: well, depending on definition of 'bleeding edge'. Maybe if some of the developers got drunk and released a bad package, you'd be more likely to bleed than if you checked out the fixes they put in the next day (or possibly the day after that) ;-) |
| 15:58.54 | *** join/#asterisk DrDeke (i=dekemar@auriaria.engin.umich.edu) |
| 15:58.57 | ManxPower | I like G726 - less bandwidth than ulaw, but still has very good quality. |
| 15:59.25 | }cytrak{ | so I can convert my wav files to G726 ? any patent issues there though ? |
| 15:59.30 | Vhata | (of course, Asterisk developers don't drink.) |
| 15:59.41 | ManxPower | }cytrak{, no free codec in Asterisk has patent issues. |
| 16:00.43 | graphyx | tmccrary: Were you able to use linphone with asterisk just fine? |
| 16:01.00 | tmccrary | yep |
| 16:01.18 | graphyx | Did you use the gnome interface or the console? |
| 16:01.32 | tmccrary | the only issues were really getting alsa to play nice with full duplex sounds (which had nothing to do with asterisk or linphone). God linux sound sucks so bad |
| 16:01.37 | tmccrary | Gnome interface |
| 16:01.47 | graphyx | any ideas on how to do it with the console? |
| 16:01.53 | graphyx | or is that something to avoid completely? |
| 16:01.54 | tmccrary | in fact, I was connecting to china, which is suprisingly useable |
| 16:02.44 | *** join/#asterisk Connor_ (n=Connor@198-144-174-5.knx.tn.nxs.net) |
| 16:02.49 | wasim | tmccrary: it doesn't ... we use alsa all the time |
| 16:03.06 | *** part/#asterisk Connor_ (n=Connor@198-144-174-5.knx.tn.nxs.net) |
| 16:03.15 | *** join/#asterisk fmano (n=fmano@195-23-20-83.net.novis.pt) |
| 16:04.00 | tmccrary | wasim: I like linux, don't get all zealoty with me, but Linux's sound support has issues |
| 16:04.09 | tmccrary | as does it's wireless, etc |
| 16:04.27 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 16:04.38 | wasim | tmccrary: i'm not getting zealoty, we use alsa for a lot more than voip, like low latency recording in studio applications etc ... i'd like to know your experience and benefit from it |
| 16:05.02 | DCGrendel | don't mix linux audio + via chipsets. crashes will ensue |
| 16:05.07 | tmccrary | I still use it on pretty much all my devices, its the only free alternative with at least decent hardware support |
| 16:05.18 | DrDeke | I have a better idea: Never use a via chipset. :( |
| 16:05.29 | DCGrendel | yes i have that opinion too |
| 16:05.43 | DCGrendel | but when the other option is to run windows on it... |
| 16:05.44 | }cytrak{ | what can be used to listen to gsm files ? |
| 16:05.51 | tmccrary | yep, there's a lot of linux bugs like that. For example, if you need opengl support with a modern chipset... don't use anything but nvidia. |
| 16:06.11 | ManxPower | }cytrak{, the "play" command that comes with sox worked for me. |
| 16:06.14 | tmccrary | but there's also a lot of upsides (like asterisk of course) |
| 16:06.18 | }cytrak{ | cool |
| 16:06.20 | }cytrak{ | thanks |
| 16:06.43 | tmccrary | cytrak as an aside, you can use sox to transcode audio files into gsm (or quite a few different formats) |
| 16:09.39 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 16:11.46 | }cytrak{ | hmmm got it converted but the gsm file doesn't seem to clear as the wav file is that normal ? |
| 16:12.05 | copantl | hi |
| 16:12.09 | }cytrak{ | checking the man sox .. brb |
| 16:12.11 | DrDeke | Yeah, a GSM file isn't going to sound as clear as a WAV file in general. |
| 16:12.28 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 16:12.52 | *** join/#asterisk johnb8989 (n=johnb898@69.60.198.133) |
| 16:14.04 | }cytrak{ | it sound like mono |
| 16:14.23 | }cytrak{ | the wav seems to have been recorded in stereo |
| 16:14.57 | copantl | i have this problem: when i dial the cli show this: Zap/1-1 is ringing |
| 16:14.57 | copantl | <PROTECTED> |
| 16:15.18 | copantl | but the other party is not answer still ringing |
| 16:15.22 | copantl | any idea? |
| 16:15.23 | ManxPower | copantl, are you calling out of an analog FXO port? |
| 16:15.35 | copantl | nop |
| 16:15.37 | copantl | PRI |
| 16:15.42 | copantl | te110P |
| 16:16.06 | ManxPower | copantl, can't imagine why unless the far end has answered and is sending back a ringing sound (line a automatic voice/fax switch) |
| 16:16.41 | DrDeke | cytrak: not much sense in using stereo files for a telephone conversation :) |
| 16:17.15 | hugov6 | how long does a with setvar made var exist? problem: i set it in label1, in label2 it works but in label3 it wont? |
| 16:17.29 | hugov6 | or better sometimes it works and sometimes not |
| 16:17.45 | copantl | here is my config diagram: |
| 16:18.40 | copantl | myasterisk----pri-----LUCENT---------anykindorcentralswitch |
| 16:20.07 | ManxPower | copantl, the only real way to tell is to do a "pri debug span x" and learn enough about PRI to diagnose the issue. I suspect the LUCENT is sending Asterisk whatever message means "call is answered" in RPISpeak |
| 16:20.18 | *** join/#asterisk Kokey (n=ubunture@201.153.63.79) |
| 16:20.52 | copantl | ok |
| 16:21.15 | copantl | ManxPower: i gonna do that and i tell you later thanx |
| 16:22.12 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 16:22.30 | X-Files | hello users, <analog phone FXS> --> <ITG> --> <ASTERISK> ->> <ITG> -->>> <analog line FXO> ..... no work dial tone .... , why i cant use dial tone ? |
| 16:23.06 | ManxPower | X-Files, ITG? |
| 16:23.26 | X-Files | Internet Telephony Gateway (2 FXO ports and 2 FXS ports) |
| 16:25.13 | X-Files | no comments ? |
| 16:25.25 | ManxPower | X-Files, sounds like you need to simplify things. |
| 16:26.10 | X-Files | hmm |
| 16:26.29 | X-Files | How? |
| 16:28.51 | brad_mssw | dialtone should be generated by your FXS device itself, so if it's not, most likely it has not registered with asterisk properly |
| 16:29.38 | X-Files | and what me doing ? |
| 16:29.52 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 16:30.34 | brad_mssw | under asterisk, do a 'sip show peers' |
| 16:30.45 | brad_mssw | does it show the ip address of your FXS device ? |
| 16:31.24 | X-Files | yep |
| 16:31.27 | Math` | yup |
| 16:31.40 | Math` | er, tought it was a question |
| 16:31.41 | Math` | lol |
| 16:31.48 | X-Files | ;) |
| 16:32.15 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 16:32.23 | cnet2 | hi everyone, |
| 16:32.32 | *** join/#asterisk Mad_Hornet (n=trodasta@70-254-74-98.ded.swbell.net) |
| 16:32.35 | brad_mssw | X-Files: can you log into your FXS device, and see if it's successfully registered with asterisk ? |
| 16:33.00 | brad_mssw | X-Files: most devices won't generate a dialtone unless there is successful registration with your server |
| 16:33.25 | cnet2 | i was wondering have you guys seen a voip phone, that looks something like those receptionist phones.. or at least have the lights that show who's line is busy. (maybe these could be programable) |
| 16:33.28 | X-Files | brad_mssw: SIP register phone 7530510 4 status: SUCCESS |
| 16:33.46 | X-Files | brad_mssw: this FXO port 1 |
| 16:33.52 | brad_mssw | X-Files: also, what's the port number you see it coming in on in your 'sip show peers' ? |
| 16:33.55 | X-Files | brad_mssw: SIP register phone 75305101 4 status: SUCCESS |
| 16:33.56 | Mad_Hornet | I need help with a TE410P installation... anyone have experience with them? |
| 16:34.00 | DrDeke | cnet2: You could look at a Grandstream 2000 : http://www.grandstream.com/y-gxp2000.htm |
| 16:34.02 | X-Files | brad_mssw: this FXS port 3 |
| 16:34.31 | cnet2 | DrDeke: thanks :P |
| 16:34.50 | X-Files | brad_mssw: check private ;) |
| 16:34.57 | DrDeke | I'm not exactly sure how you would go about programming the buttons though. |
| 16:34.59 | brad_mssw | X-Files: I'll reply there |
| 16:35.11 | X-Files | yep |
| 16:35.26 | graphyx | can sjphone do multiple lines? |
| 16:35.40 | tmccrary | cnet, you may want to check the snom 360 with the attachable switch board |
| 16:36.03 | tmccrary | Mad_Hornet: I have a TE410P |
| 16:37.34 | Mad_Hornet | tmccrary: do you have a full T1? We have 4 channels split off for voice, I can see where the telco company punched them down on the 66 block, but I am not sure where to go from here... New to telephony... |
| 16:39.00 | cnet2 | tmccrary: ok.. thx |
| 16:40.47 | cnet2 | that snom 360 phone looks awsome |
| 16:40.57 | wunderkin | Mad_Hornet: you can see what? where the 4 voice lines are punched down? |
| 16:43.50 | Mad_Hornet | wunderkin: I can see the 2 pair split from the T1 line punched down on the 66 block... |
| 16:44.05 | marcus2 | so make a cable and plug it into your te410p |
| 16:44.21 | marcus2 | why didn't the telco give you a rj45 jack? thats kind of odd |
| 16:44.30 | *** join/#asterisk seong (i=seong@60.49.70.190) |
| 16:44.43 | DrDeke | Hey, this is kind of off-topic, but do any of you happen to know if a T3/DS3 comes into a premesis via copper or fiber? If copper, 2 pairs like a T1? Or something else? |
| 16:44.56 | ManxPower | marc324, E-1s are commonly handed off as 2 coax cables, there are adapters to convert to RJ-45 |
| 16:44.59 | DrDeke | premises* |
| 16:45.05 | marcus2 | DS3s come over 2 coax lines, or fiber. |
| 16:45.06 | Math` | http://cgi.ebay.com/Packet8-Broadband-VOIP-Phone-Adapter-best-of-the-best_W0QQitemZ5828342531QQcategoryZ61841QQrdZ1QQcmdZViewItem#ebayphotohosting |
| 16:45.06 | [TK]D-Fender | cnet2 : take a look at the Polycom 601 & attendant modules.... |
| 16:45.14 | Math` | any ways to flash that to work with asterisk? |
| 16:45.41 | ManxPower | Math`, Don't buy used Vendor Locked ATAs. |
| 16:45.44 | marcus2 | manx; he said they are two pairs, not two bits of coax. |
| 16:46.06 | Math` | ManxPower: except if you know how to unlock them lol |
| 16:46.11 | Mad_Hornet | marcus2: I think so too... to make a cable I would just punch down over the 2 pair? I haven't done this before... |
| 16:46.17 | ManxPower | Math`, Still not worth it. |
| 16:46.30 | Math` | depends what brand |
| 16:46.35 | *** join/#asterisk bweschke (n=bweschke@101.sub-70-209-10.myvzw.com) |
| 16:46.43 | ManxPower | Yo can buy a 2-port Sipura ATA for $70 |
| 16:46.45 | marcus2 | yeah, one pair goes to pins 1/2 in the rj45, the other pair goes to 4/5 |
| 16:47.02 | brettnem | DrDeke: DS3 is delivered to premises via fiber. typically handed off via 2 coax cables |
| 16:47.05 | marcus2 | if you don't get carrier, try wiring up the pairs the other way :) |
| 16:47.19 | InfraRed | DS3 sucks |
| 16:47.23 | marcus2 | you can buy a 2 port linksys-branded sipura for $50 =D |
| 16:47.31 | brettnem | what? |
| 16:47.34 | wunderkin | Mad_Hornet: i haven't either, but they are supposted to go on the opposite side |
| 16:47.45 | DrDeke | What's wrong with DS3s? What if you need more than a T1 or two's worth of DS0s? |
| 16:47.48 | wunderkin | i think the customer side is the left normally |
| 16:48.05 | brettnem | DrDeke: there is nothing wrong with a DS3.. really |
| 16:48.08 | Math` | marcus2: where? :P |
| 16:48.14 | DrDeke | hehe |
| 16:48.17 | marcus2 | circuit city, staples, frys, walmart |
| 16:48.21 | marcus2 | lots of places online |
| 16:48.27 | InfraRed | DS3 sucks today, something must suck, DS3 it is for today \o/ |
| 16:48.32 | DrDeke | lol, ok :) |
| 16:48.39 | marcus2 | linux sucks more than DS3s ;) |
| 16:48.46 | DrDeke | Do they even make DS3 line cards that work with Asterisk? ;) |
| 16:48.57 | DrDeke | man, get off the shed, there is nothing wrong with linux ;) |
| 16:48.57 | Math` | digium's working on it |
| 16:48.58 | InfraRed | linux bashing is sooooooo 1999 |
| 16:49.00 | brettnem | DrDeke: no.. digium likes to talk about doing it tho |
| 16:49.02 | InfraRed | it's OSX days |
| 16:49.03 | InfraRed | :) |
| 16:49.05 | Math` | lol |
| 16:49.09 | [TK]D-Fender | Sangoma already has a DS3 card.... |
| 16:49.09 | InfraRed | OSX is \SO\ gay |
| 16:49.17 | Math` | osx isn't gay |
| 16:49.22 | brettnem | I don't think it's channelized |
| 16:49.35 | DrDeke | Wait, so you can't use it for voice? |
| 16:49.42 | DrDeke | (as in PSTN) |
| 16:49.47 | brettnem | that is a loaded question |
| 16:50.02 | brettnem | but your assumption is correct if mine is as well |
| 16:50.11 | DrDeke | I mean it doesn't really matter; I am not going to be getting a DS3 any time soon (ever)... Just curious is all :) |
| 16:50.33 | brettnem | well don't count on digium actually doing the DS3 |
| 16:50.43 | brettnem | and if they do, don't count on it working [well] |
| 16:50.46 | DrDeke | lol |
| 16:50.47 | Math` | lol |
| 16:50.49 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 16:50.59 | brettnem | it'll be like a DS3 card, but you can only actually use the first 4 DS1s |
| 16:51.16 | Math` | out of 27 :P |
| 16:51.19 | DrDeke | lol |
| 16:51.21 | brettnem | or maybe the zaptel channels only go to 256 or something stupid like that |
| 16:51.25 | marcus2 | my te410p seems to work pretty well.. |
| 16:51.39 | brettnem | Math`: out of 27? <sic> I suppose? |
| 16:51.43 | DrDeke | I miss my days of dealing with ISDN |
| 16:51.52 | brettnem | DrDeke: PRI or BRI? |
| 16:51.59 | DrDeke | BRI at home, PRI at work |
| 16:52.02 | Math` | brettnem: a DS3 is 27 DS1 no? |
| 16:52.06 | tzanger | 28 |
| 16:52.09 | brettnem | no, it's 28.. |
| 16:52.12 | Math` | ah ok |
| 16:52.26 | brettnem | but I found it humorous anyway.. digium might make a 27 channel DS3.. haha |
| 16:52.33 | Math` | lol |
| 16:52.35 | Mad_Hornet | I will give it a try. Another quick question... I have two POTS lines as well, can the TE410P handle them too? I read that it does FXS/FXO, but wasn't sure if it can have POTS lines plugged in. |
| 16:52.47 | brettnem | in which you can only use the first 4 channels.. |
| 16:52.50 | stevie20 | cya |
| 16:53.11 | brettnem | you cannot plug analog phone lines into T1 interfaces |
| 16:53.12 | marcus2 | sure, you can plug 4 pots lines into a te410p, as long as you have a channel bank in the middle |
| 16:53.23 | Math` | lol |
| 16:53.29 | brettnem | btw, don't even try it.. ring voltage could fry the card |
| 16:53.41 | Math` | you can plug a POTS into an Ethernet switch, as long as you have an ATA in the middle |
| 16:53.56 | marcus2 | true, true |
| 16:54.03 | [TK]D-Fender | </stupid> !!!! |
| 16:54.17 | brettnem | I can drive my car to antartica as long as there is a boat in the middle. HAR HAR.. ok, enough really |
| 16:54.22 | DrDeke | lol |
| 16:54.24 | Math` | lol |
| 16:54.51 | ManxPower | I can have my Digium card share interrupts if...um...OK, so I can't. |
| 16:54.59 | brettnem | haha |
| 16:55.12 | DrDeke | hahaha |
| 16:55.13 | brettnem | remember bkw_'s analogy of that? |
| 16:55.25 | ManxPower | brettnem, no |
| 16:55.39 | brettnem | digium cards share interrupts like a fat kid in a candy store |
| 16:55.55 | marcus2 | why should they need to share interrupts |
| 16:56.04 | brettnem | oh sheesh.. did I really say that.. shame on me |
| 16:56.10 | brettnem | marcus2: they don't NEED to.. haha |
| 16:56.31 | brettnem | you just power up and it's on the same interrupt as.. say, your SATA card.. or your ethernet card.. |
| 16:56.42 | Math` | lol |
| 16:56.44 | lme | hem |
| 16:56.48 | lme | stupid question |
| 16:56.51 | [TK]D-Fender | An E1000 no less. |
| 16:56.52 | brettnem | I want a nice DS0 bus on digital interfaces.. |
| 16:56.58 | Math` | lme: go ahead, lots of stupid stuff going on anyways |
| 16:56.59 | brettnem | like old skool SC-BUS |
| 16:57.08 | brettnem | yay stupid questions |
| 16:57.14 | lme | is it possible to continue in dial plan after a Dial command and if caller hang before called |
| 16:57.15 | lme | ? |
| 16:57.30 | marcus2 | my systme has more free interrupts than it knows what to do with ! |
| 16:57.36 | ManxPower | lme, "show application dial", pay special attention to the "g" option |
| 16:57.38 | Math` | you mean if you dial and you hangup before having finished dialing? |
| 16:57.56 | brettnem | ManxPower: I think he means if the callER hangs up |
| 16:57.59 | [TK]D-Fender | marcus2 : Mine had TONS of free interrupts, but would always try and chare with my TDM400.... |
| 16:58.07 | marcus2 | sounds like a crappy bios |
| 16:58.15 | ManxPower | brettnem, that would be exten => h of course. |
| 16:58.17 | lme | ManxPower: it's only working if the called hang, not the caller |
| 16:58.21 | brettnem | heh..dosen't matter how many free interrupts you have.. crap bios will assign them all to the same ones. |
| 16:58.36 | brettnem | ManxPower: does that work these days.. everone always cautioned against using h |
| 16:58.45 | [TK]D-Fender | marcus2 : happens on a lot of boards. My Gigabyte one had the ability to reserve by slot and didn't seem to work, and thats through a BIOS upgrade as well |
| 16:58.58 | brettnem | [TK]D-Fender; same here |
| 16:58.58 | ManxPower | brettnem, it's a bad idea, but it works |
| 16:59.00 | marcus2 | most motherboards are crap :) |
| 16:59.02 | *** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 16:59.08 | marcus2 | use decent boards, you wont have problems :) |
| 16:59.09 | brettnem | ManxPower: ah ok, just wanted to get that out.. ;) |
| 16:59.13 | ManxPower | If you want to dial and then hangup, use a damn .call tile. |
| 16:59.20 | [TK]D-Fender | <3 Sangoma <3 |
| 16:59.49 | brettnem | good point |
| 16:59.58 | Sobakai | can someone help me with my FOP its acting rather weird... |
| 17:00.02 | *** part/#asterisk Mad_Hornet (n=trodasta@70-254-74-98.ded.swbell.net) |
| 17:00.05 | brettnem | I think sangoma will save us all.. but I don't think it'll be part of the asterik project.......... |
| 17:00.08 | marcus2 | i'm always amused when people slap a $1500 pci card in a $75 cheap-ass motherboard and then bitch that the pci card isnt compatible with the mobo |
| 17:00.20 | *** join/#asterisk Timoti (n=asqsa@85.99.166.94) |
| 17:00.26 | Math` | brettnem: except if zomeone writes a sap driver for it |
| 17:00.30 | Math` | er, someone, zap |
| 17:00.46 | Timoti | Hi everybody |
| 17:00.49 | Timoti | I need your help |
| 17:00.50 | ManxPower | Math`, Um, Sangoma already has Zap drivers for many of their boards |
| 17:01.01 | lme | ManxPower: in fact i want to tag my cdruserfield with the correct state, answer, lost, voicemail, but if caller hang before called, it's stop in the dialplan and exit non-zero... |
| 17:01.01 | [TK]D-Fender | marcus2 : and at the same time ask yourself why such a HUGE % of server MB's use E1000 NIC's, and Digium's cards can't cope? |
| 17:01.04 | brettnem | zap sux.. |
| 17:01.11 | marcus2 | uhm? |
| 17:01.13 | brettnem | really.. chan_sangoma.. hellllooooOooo! |
| 17:01.15 | Timoti | I just install asterisk @home .. and now I would like to add H323 addon ... |
| 17:01.17 | ManxPower | lme, Then you need to do that in exten => h |
| 17:01.26 | marcus2 | i have a te410p in a machine with two e1000s |
| 17:01.30 | marcus2 | seems to work just fine to me? |
| 17:01.32 | Timoti | I red the instruction ... and I have no idea related linux |
| 17:01.40 | Timoti | at the instruction it is saying |
| 17:01.41 | Timoti | Copy the asteriskathome-h323.zip file to you Asterisk@Home server using WinSCP. Unzip the file by typing |
| 17:01.48 | Timoti | How can I do that ? |
| 17:01.55 | marcus2 | you need to go ask on #unix |
| 17:01.57 | Qwell | By typing what it...tells you...to...type |
| 17:02.04 | Timoti | to which directory .. and by which commands ? |
| 17:02.07 | ManxPower | marcus2, I hear over and over that you need to disable many onboard 1Gbps Ethernet devices to get Digium cards to work correctly. |
| 17:02.14 | Qwell | Timoti: the ones they give you in the next line |
| 17:02.19 | }cytrak{ | hmm do I need to created an AGI to receive input from a user ? |
| 17:02.21 | marcus2 | manx; dual onboard e1000s, no problems |
| 17:02.27 | ManxPower | Timoti, We can't help you with Asterisk@Home. |
| 17:02.36 | ManxPower | marcus2, For YOU, but others have had problems. |
| 17:02.37 | marcus2 | but ... i'm using a decent motherboard. |
| 17:02.49 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:02.49 | [TK]D-Fender | }cytrak{ : nope you can use the Read command to get inoput from the user. |
| 17:02.51 | Timoti | well I know .. but can you help me ? |
| 17:02.58 | file | we use e1000s, compiled into the kernel... that works fine |
| 17:02.59 | marcus2 | he just said we can't |
| 17:03.00 | brettnem | Timoti ~gwypf |
| 17:03.03 | }cytrak{ | I want a user to enter for example extension number , i want to be able to pick that up from the dialplan and save in a channel var |
| 17:03.07 | Timoti | or do you hate asterisk@home ... so you hate me too ? |
| 17:03.08 | ManxPower | Timoti, I can't. I've never even been to the Asterisk@home web site. |
| 17:03.12 | brettnem | ~gwypf |
| 17:03.13 | jbot | it has been said that gwypf is Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 17:03.14 | Qwell | Timoti: yes, pretty much |
| 17:03.14 | [TK]D-Fender | }cytrak{ just depends what you want to do with it after that will decide if AGI is the BEST thing to do in your case |
| 17:03.16 | }cytrak{ | cool thanks .. forgot about read |
| 17:03.18 | Qwell | bbl, work |
| 17:03.28 | marcus2 | my asterisk server has e1000 loaded as a module, still no problems |
| 17:03.28 | lme | ManxPower: ok... that's the point... thanx |
| 17:03.32 | [TK]D-Fender | }cytrak{ : "Read" is your answer then. |
| 17:03.35 | Timoti | :-(( |
| 17:03.35 | file | [TK]D-Fender: I see you |
| 17:03.42 | [TK]D-Fender | :O |
| 17:03.54 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 17:03.57 | file | darn, now I don't |
| 17:04.05 | }cytrak{ | so i do a simple playback (enter extension number) and then call read |
| 17:04.07 | brettnem | Timoti: really.. try to follow the directions before asking.. then go ask in the A@H fourms |
| 17:04.18 | *** part/#asterisk Timoti (n=asqsa@85.99.166.94) |
| 17:04.27 | *** join/#asterisk expressfone1 (n=expressf@62-15-97-163.inversas.jazztel.es) |
| 17:04.29 | expressfone1 | Hi |
| 17:04.29 | DrDeke | what's chan_cloak? |
| 17:04.37 | marcus2 | hrm. "This seems to be an issue with the e1000 and not with the Zaptel device." |
| 17:04.49 | [TK]D-Fender | DrDeke : if you can't see it it must be working! |
| 17:04.52 | DrDeke | lol |
| 17:05.18 | file | woot clean glasses |
| 17:05.20 | [TK]D-Fender | Ok, I'm off for lunch.... |
| 17:05.22 | expressfone1 | can * run on MIPS-32 4Kc procesor?? |
| 17:05.24 | file | [TK]D-Fender: now I see you even better |
| 17:05.31 | brettnem | ah ha.. i must have that one loaded too.. right next to res_crash_once_a_week.so |
| 17:05.42 | marcus2 | i run * on a mips cpu. dunno if its a 4kc |
| 17:05.49 | brettnem | and res_memory_leak.so |
| 17:05.59 | file | don't forget res_toaster.so for all your toasting needs |
| 17:06.04 | expressfone1 | marcus2> with zaptel cards?? |
| 17:06.11 | brettnem | res_buggy.so |
| 17:06.24 | marcus2 | nope, no zap cards |
| 17:06.26 | Math` | I beleive res_memory_leak.so takes an "mb/hour" argument |
| 17:06.27 | marcus2 | no pci on that system |
| 17:06.33 | Sobakai | Can anyone help me with my Flash Op Panel, its acting rather weird... |
| 17:07.33 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:07.34 | expressfone1 | marcus2> what mips board you use?? |
| 17:07.43 | marcus2 | if you all hate digium and asterisk so much, why do you hang out here? =D |
| 17:07.49 | DCGrendel | heh |
| 17:07.49 | marcus2 | express; a linksys wrt54gs |
| 17:08.52 | Rawplayer | re |
| 17:08.58 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 17:10.10 | }cytrak{ | hm when you use read it creates a channel var or I have to create that channel var with setvar before calling read ? |
| 17:10.22 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 17:11.18 | Sobakai | is there a Flash Panel Op channel I can join maybe? |
| 17:11.29 | brettnem | marcus2: because openpbx isn't developed yet. ;) |
| 17:11.45 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:11.58 | marcus2 | heh |
| 17:12.03 | marcus2 | why arent you working on it/1 |
| 17:12.04 | marcus2 | ?! |
| 17:12.15 | brettnem | I'm a user |
| 17:12.17 | brettnem | :) |
| 17:12.31 | brettnem | Trust me, if I knew more about coding; I'd be all over it |
| 17:13.27 | *** join/#asterisk grimse (n=grimse@p5481F350.dip.t-dialin.net) |
| 17:13.43 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-137-217.37-151.net24.it) |
| 17:14.44 | brettnem | oh.. lunchtime |
| 17:14.52 | *** part/#asterisk Gh0sty (n=ghosty@ip-81-11-227-234.dsl.scarlet.be) |
| 17:15.06 | lme | ManxPower: well, ${CDR(dst)} = h in this case :) |
| 17:15.34 | wunderkin | where art thou cresl1n :( |
| 17:17.59 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 17:18.16 | kpettit | is there any tooks to query Zap channels to make sure there working? |
| 17:19.05 | kpettit | I'm getting alot of zaptel fx problems, and from asterisk I can't see that there are issues, but if I restart asterisk which reloads the zaptel modules it fixes everytyhing. It's just anoying becuase i have to do it on most machines at least once a month |
| 17:19.18 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 17:20.42 | [TK]D-Fender | }cytrak{ : It should create the channel variable if it doesn't exist |
| 17:21.05 | *** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net) |
| 17:21.57 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 17:22.09 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 17:23.09 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 17:24.13 | ManxPower | kpettit, define "fx errors" |
| 17:24.30 | ManxPower | or "fx problem" |
| 17:25.31 | *** join/#asterisk Wunsch (n=scott@tachyon.ip4.wunsch.org) |
| 17:26.04 | *** part/#asterisk graphyx (n=mike@67.50.46.118) |
| 17:27.39 | kpettit | I have analog faxes or paging systems. That basically quit working. |
| 17:28.22 | kpettit | The paging system just gets horrible static noise, faxes can't send. I've even had problem with stobe lights I've hookd up to analog ports on the fx cards |
| 17:28.37 | kpettit | in all these problems a asterisk restart which reloads the zaptel module fixes the problems. |
| 17:29.15 | kpettit | The annoying thing is when I go into asterisk there is no indication there is a problem. With the stobe light I was doing it would show the zap channel as busy all the time, but that's about it |
| 17:31.10 | kpettit | I've got this problem on pretty much every machine I have with a zaptel fx card. |
| 17:34.06 | InfraRed | kpettit: what cards are you using |
| 17:34.18 | tmccrary | kpettit have you filed a bug report |
| 17:34.53 | [TK]D-Fender | Please clarify "fx" card. specify a real model since "fx" doesn't mean much |
| 17:35.11 | tmccrary | intel modem ;) |
| 17:35.49 | ManxPower | kpettit, you mean FXS ports. |
| 17:35.58 | }cytrak{ | could some check this out for me please http://pastebin.com/430602 |
| 17:36.06 | ManxPower | kpettit, make sure your Digium cards are not shareing interrupts by using: cat /proc/interrrupts |
| 17:36.41 | *** join/#asterisk Kort (n=james@65.211.216.202) |
| 17:36.49 | }cytrak{ | I used Read as mentioned but even though I entered a number I got user entered nothing in the asterisk console |
| 17:36.55 | Kort | does Asterisk have a variable that contains the extension the call originated from? |
| 17:37.09 | }cytrak{ | yes |
| 17:37.20 | ManxPower | exten => 8000,10,Gotoif($["${train}" = "21"]?11:12) or # |
| 17:37.21 | ManxPower | exten => 8000,10,Gotoif($[X${train} = X21]?11:12) |
| 17:37.22 | }cytrak{ | I think its $EXTEN |
| 17:37.31 | Kort | $EXTEN is the dialed number |
| 17:37.41 | }cytrak{ | opps sorry |
| 17:37.44 | }cytrak{ | true |
| 17:37.59 | ManxPower | }cytrak{, I suspect if TRAIN is empty you'll get that error since your existing test would evaluage to " = 21" |
| 17:38.15 | X-Files | ppl please help . i connect from asterisk to FXO and i have dialtone , i want in num pad enter phone number. tone number not accept and this dialtone not changed to call |
| 17:38.15 | }cytrak{ | I'm sure there is other wise you wouldn't be able to use the exgirlfriend stuff |
| 17:38.20 | Kort | I'll just parse $CALLERID then.. |
| 17:38.43 | }cytrak{ | ManxPower: I c |
| 17:38.44 | X-Files | where problem ? asterisk or my FXO port configure ? |
| 17:38.44 | ManxPower | X-Files, We can't help you because your system is too complex. Read up on the Asterisk docs |
| 17:38.46 | ManxPower | ~docs |
| 17:38.47 | jbot | docs is, like, 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 |
| 17:38.59 | ManxPower | }cytrak{, I don't know WHY TRAIN is empty, but..... |
| 17:39.21 | ManxPower | X-Files, You need to make the ANALOG PHONE -> ADAPTER -> Asterisk work first. |
| 17:39.35 | ManxPower | You should get dialtone from the adapter, if you don't, then that's what you need to fix first. |
| 17:39.49 | X-Files | ManxPower: analog phone ->> adapter --> asterisk work ! |
| 17:39.58 | ManxPower | This is not your own personal help channel. We are doing this stuff for free. If you want someone to hold your hand then hire an Asterisk consultant |
| 17:40.09 | kpettit | ManxPower, it's not sharing the interrupts |
| 17:40.14 | X-Files | ManxPower: analog phone FXO ->> adapter --> asterisk work too |
| 17:40.17 | }cytrak{ | ManxPower: you think it's because I forgot the quotes ? |
| 17:40.41 | ManxPower | X-Files, so you can call extensions (like voicemail) on the Asterisk system from the analog phone? |
| 17:40.49 | ManxPower | }cytrak{, yes. |
| 17:41.00 | X-Files | ManxPower: but from call analog phone FXS -->> adapter --> ASTERISK ---> Adapter --> FXO (dialtone have) but i can't send tone number for call |
| 17:41.15 | X-Files | ManxPower: yes, voicemail work perfect |
| 17:41.16 | ManxPower | Try my example. The 2nd example I gave is an old shell scripting trick for dealing with tests for empty variables that also work in Asterisk |
| 17:41.29 | kpettit | InfraRed, tmccrary I'm using the quad 400p digium card |
| 17:41.46 | ManxPower | X-Files, so now you need to figure out why the ASTERISK -> ADAPTER -> PHONE LINE is not working. |
| 17:41.47 | kpettit | we have talked to digum but they basically deny that there card has anyissues |
| 17:41.56 | ManxPower | If you plug your analog phone directly into the PHONE LINE does it work? |
| 17:42.26 | kpettit | nothing I plug directlyh into the port will work once it's hung. I have to restart asterisk for it to work |
| 17:42.26 | ManxPower | kpettit, I have something like 6 or so Digium cards in use, including at least one or 2 4-port cards and have never seen the problem you are experiencing. |
| 17:42.52 | kpettit | I've got over 40 of those 4 port cards. They drive me freaking nuts |
| 17:43.07 | X-Files | ManxPower: ASTERISK -> ADAPTER -> PHONE LINE FXO is worked ! but i can't send DIAL NUMBER (example 5455545) |
| 17:43.08 | ManxPower | kpettit, I assume you did the standard stuff, like download latest version of the branch of Asterisk you are using (1.0.9 or 1.2RC2) for Zaptel, libpri, and Asterisk? |
| 17:43.22 | kpettit | for plain analog phone they don't seem to have issues. but with fax, paging system, strobes I get issues where I have to restart every so often |
| 17:43.29 | ManxPower | X-Files, put the Asterisk console output of a failed call on pastebin.ca |
| 17:43.32 | kpettit | 1.0.9 |
| 17:43.40 | X-Files | ManxPower: ok wait |
| 17:43.53 | kpettit | I'm waiting for the stable of 1.2 so I can do the real-time stuff. I'm very excited about that stuff |
| 17:44.15 | *** join/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca) |
| 17:44.23 | kpettit | ManxPower, that's the problme. When I ring a long it just shows it rining Zap/1-1 or whatever that's it |
| 17:44.26 | ManxPower | kpettit, Well, 1.2 has some significant bug fixes, but I ran a 4-port card using all 4 ports using 1.0.7 - 1.0.9 with no problems |
| 17:44.43 | X-Files | ManxPower: http://pastebin.ca/28821 |
| 17:44.47 | ManxPower | kpettit, Oh! Is that the quad T-1 card or the quad ANALOG card? |
| 17:44.48 | kpettit | can't tell there's a issue unless I get a call that "fax isn't working" or "the paging light isn't working" |
| 17:45.02 | kpettit | quad analog |
| 17:45.18 | kpettit | I've got some quad t-1 cards too. those kick major ass |
| 17:45.26 | kpettit | no problem at all with those |
| 17:45.36 | }cytrak{ | ManxPower: I used your example + initialized the var to 0 in the begining of the dialplan but it just seems like what I'm entering on the DMTF dialpad doesn't get noticed. |
| 17:45.38 | ManxPower | kpettit, Oh! Yes, those cards suck. I try not to use them. They lock up on my systems (where I still have them) about once per month or so, depending on usage. |
| 17:45.45 | *** join/#asterisk cyberkoa (n=cyberkoa@65.90.93.3) |
| 17:45.55 | }cytrak{ | I had a problem similar to that before I had to change the dmtf to rfc |
| 17:45.57 | kpettit | ManxPower, Exactly!! Once a month seems to be it |
| 17:45.59 | ManxPower | }cytrak{, you may have a DTMF problem and not a dialplan problem |
| 17:46.03 | *** part/#asterisk cyberkoa (n=cyberkoa@65.90.93.3) |
| 17:46.04 | }cytrak{ | but I'm already using rfc |
| 17:46.13 | ManxPower | kpettit, There is no good fix. |
| 17:46.14 | }cytrak{ | using kphone by the way |
| 17:46.22 | }cytrak{ | let me try with x-lite |
| 17:46.25 | kpettit | I'd just like to find a way to test, but I'm not sure how to do that. I don't really want ot do the "reboot once a week" thing |
| 17:46.33 | ManxPower | }cytrak{, remember the phone and asterisk can't tell what the other side is using, so they have to be both forced to rfc2833 |
| 17:46.57 | ManxPower | kpettit, the problem is that we can't reproduce the problem so we can't get Digium to fix the problem. |
| 17:46.57 | X-Files | ManxPower: line 53 in http://pastebin.ca/28821 , dialtone from FXO port , i enter phone number from num pad , line 58 i hangup line ... |
| 17:47.12 | ManxPower | kpettit, My solution is to get a 1 or 2 port T-1/E-1 card and use a channelbank. |
| 17:47.20 | ManxPower | X-Files, you are missing /etc/asterisk/indications.conf |
| 17:47.43 | kpettit | ManxPower, I'd love to do that, but they are two expensive. |
| 17:47.45 | X-Files | hmm |
| 17:48.07 | kpettit | I need to start expermenting with hardware. This are getting reallyanoying |
| 17:48.11 | ManxPower | kpettit, I know. Less expensive than you might think. |
| 17:48.33 | *** join/#asterisk mxmasster (n=mxmasste@ppp-71-140-105-24.dsl.irvnca.pacbell.net) |
| 17:48.33 | kpettit | the t-1 cards are over 1k arent they? |
| 17:48.34 | mxmasster | hi all |
| 17:48.34 | X-Files | ManxPower: i have this file |
| 17:48.38 | ManxPower | X-Files, Also your Dial line is wrong. You are telling it to dial extension 201 off the FXO device |
| 17:48.39 | kpettit | the PRI t-1 type cards? |
| 17:48.54 | kpettit | they owner buys all the hardware. I just get it working |
| 17:48.54 | mxmasster | i am looking for instructions on how to create RPMS for an asterisk install on my RedHat systems |
| 17:49.06 | ManxPower | kpettit, 1-port is $500 + eBay Adtran ChannelBank $200 - $350 |
| 17:49.16 | *** join/#asterisk YoYo (n=yoyo@carter.psknet.com) |
| 17:49.16 | kpettit | that's not too bad. |
| 17:49.24 | ManxPower | mxmasster, why don't you look at how the existing RPMS for Redhat are built? |
| 17:49.29 | kpettit | these 4 ports are sooo perfect if they would just work |
| 17:49.44 | X-Files | ManxPower: in extensions.conf exten => 1,1,Dial(SIP/201@85.115.115.125) |
| 17:50.01 | kpettit | have you tried any other multi port fx type cards? |
| 17:50.03 | X-Files | ManxPower: why this line is wrong ? |
| 17:50.05 | ManxPower | kpettit, I know. Digium doesn't admit there's a problem, users know there's a problem. As I said, my fix is to just replace the analog cards with T-1 cards and a channel bank |
| 17:50.25 | kpettit | digium is funny like that. |
| 17:50.35 | ManxPower | X-Files, exten => 1,1,Dial(SIP/telephonenumberyouwanttodial@sipconfentry) |
| 17:50.59 | ManxPower | kpettit, Of course, you never see the problem until it's too late to return the cards and buy Sangoma |
| 17:51.04 | ManxPower | or Voicertonix |
| 17:51.16 | kpettit | no kidding. After we've bought like a 100 of them |
| 17:51.33 | X-Files | ManxPower: yep, in ADAPTER configured local phone 201 = FXO port |
| 17:51.33 | kpettit | have you tried Sangoma? |
| 17:51.47 | ManxPower | X-Files, Make sure your analog adapters have Silence Supression DISABLED and that the DTMF mode for BOTH the devices AND Asterisk is set to rfc2833 |
| 17:52.03 | ManxPower | kpettit, no, but many people love them |
| 17:52.03 | }cytrak{ | I think there is something wrong with how I'm setting the var with Setvar(train=0) and how Read sets the train var .. they don't seem to be the same |
| 17:52.19 | ManxPower | kpettit, the problem is there is not a lot of support from the community for Sangoma |
| 17:52.24 | *** part/#asterisk Moc- (n=mochouin@modemcable181.215-82-70.mc.videotron.ca) |
| 17:52.38 | ManxPower | X-Files, then you have a problem with your analog adapter, not Asterisk. |
| 17:52.48 | ManxPower | X-Files, What brand/model of analog adapter are you using? |
| 17:52.50 | X-Files | ManxPower: silence (VAD) disabled in ADAPTER |
| 17:53.03 | wasim | its simpler to just buy 4 port fxs sip ata |
| 17:53.19 | ManxPower | X-Files, No, it's not. This message means the adapter is using VAD/CNG/etc "Nov 15 19:36:58 NOTICE[13001]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible" |
| 17:53.22 | *** join/#asterisk cpatry (n=grepmoo@65.39.228.5) |
| 17:53.28 | kpettit | ManxPower, I think I'll give them a try and just test the hell out of them to see how they do |
| 17:53.40 | ManxPower | wasim, or two 2-port SIPura SAP2001 |
| 17:53.50 | X-Files | ManxPower: ADAPTER Eusso UTG7104-22 |
| 17:53.54 | ManxPower | kpettit, consider a SIP ATA, they are cheaper. |
| 17:53.59 | ManxPower | X-Files, I cannot help you with that device. |
| 17:54.13 | YoYo | since updating my cisco 7940's to 7.5, we're getting a weird double-ring. I set progressinband=never, but it's still the same |
| 17:54.22 | X-Files | ManxPower: Planet VIP-000/400/420 ? |
| 17:54.22 | YoYo | anyone have any suggestions? |
| 17:54.30 | ManxPower | 2 x SPA-2001s should cost US$120 - 140 |
| 17:55.05 | ManxPower | X-Files, I have only used Cisco and SIPUra ATAs and most people stick to those or the crappy Grandstream devices. I've never hear of either of your devices. |
| 17:55.23 | ManxPower | kpettit, the problem is that Fax over SIP doesn't work very well. |
| 17:55.37 | ManxPower | kpettit, your best solution is reboot the server every 2 weeks until you can find a long term solution |
| 17:55.58 | X-Files | ManxPower: eh :( dtmf_relay turn is on |
| 17:56.03 | kpettit | We have a pretty good setup becuase we have direct p2p with the ISP which is also a sip provider so it works prettywell |
| 17:56.17 | kpettit | but I've noticed on certain fax resolutions work. |
| 17:56.38 | ManxPower | X-Files, DTMF Relay may be RFC2833, and it needs to be set. |
| 17:57.12 | X-Files | ManxPower: i configure last time dtmf relay all profile turn is on ... |
| 17:57.24 | X-Files | old last time :) |
| 17:57.40 | X-Files | brr |
| 17:57.43 | X-Files | paste ;( |
| 17:57.43 | ManxPower | X-Files, The only thing I can suggest is for your to search the mailing lists or post your question on the mailing lists, someone may have the same device |
| 17:57.45 | ManxPower | ~mailinglist |
| 17:57.47 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 17:58.14 | X-Files | i checked this site :( not found :( |
| 17:58.46 | X-Files | maybe i lame :( but no result eusso |
| 17:59.42 | zoa | manxpower |
| 17:59.58 | zoa | http://www.asteriskguru.com/archives/search.php |
| 18:00.00 | zoa | try this search |
| 18:00.15 | zoa | its realtime, not depending on if google spidered it or not |
| 18:00.27 | YoYo | since updating my cisco 7940's to 7.5, we're getting a weird double-ring. I set progressinband=never, but it's still the same. how do I track down where the ringing is coming from? |
| 18:01.03 | ManxPower | zoa, well search author (my e-mail address) healds no hits |
| 18:01.20 | zoa | ah, its because it does at |
| 18:01.31 | zoa | try something else like dtmf |
| 18:01.34 | *** join/#asterisk razu_ (n=razu@ip201.cab19.mus.starman.ee) |
| 18:01.41 | zoa | there is some anti spam stuff on the email address |
| 18:01.54 | ManxPower | zoa, search on my last name has no results |
| 18:02.42 | ManxPower | zoa, search for my domain in author field fails |
| 18:02.48 | zoa | sarch for author doesnt seem to work indeed |
| 18:03.00 | zoa | search for a topic |
| 18:03.28 | zoa | hmm i will need to fix that author thing, that sucks |
| 18:04.13 | ManxPower | zoa, Sometimes I know WHO wrote something, but not WHAT they wrote 8-) |
| 18:05.01 | zoa | yeah i know that |
| 18:05.04 | }cytrak{ | anything special has to be done to reset a value of a variable in the dial plan ? |
| 18:05.15 | zoa | stupid thing, will try to get it fixed soon |
| 18:05.17 | ManxPower | }cytrak{, no. |
| 18:05.27 | zoa | im low on php guru's so it might take a while |
| 18:05.31 | ManxPower | zoa, you know I always try the wierd stuff first. |
| 18:05.41 | zoa | hehe yeah |
| 18:05.45 | zoa | FREAK! |
| 18:05.46 | zoa | :) |
| 18:06.11 | zoa | ManxPower: are you coming to eu now ? |
| 18:06.22 | ManxPower | My last testbed system SIP->Asterisk->GRE Tunnel->Asterisk->Tellabs EchoCan-> Asterisk->SIP->Analog phone. |
| 18:06.43 | ManxPower | zoa, nobody offered me a job so I'm going to live in the mountians |
| 18:07.04 | ManxPower | leaving thursday evening to do a site survey for internet access |
| 18:07.31 | zoa | in the mountains ? |
| 18:07.32 | zoa | why ? |
| 18:07.37 | ManxPower | zoa, why not? |
| 18:07.43 | wasim | earthquakes |
| 18:07.46 | zoa | but what are you going to find int he mountains ? |
| 18:07.49 | ManxPower | wasim, not in these |
| 18:07.57 | X-Files | ManxPower: exten => 201,1,Dial(SIP/201@85.115.115.125) <--- this correct ? |
| 18:08.01 | zoa | wild bears ! |
| 18:08.07 | stbain | bliss... I love living in the mountains |
| 18:08.07 | zoa | no fresh water |
| 18:08.08 | *** join/#asterisk fugitivo (n=ajf@209.13.244.248) |
| 18:08.12 | ManxPower | zoa, It's what I WON'T find in the mountians - crime, people, smog, noise. |
| 18:08.12 | zoa | mosquito's |
| 18:08.15 | drumkilla | zoa: your site rocks |
| 18:08.17 | zoa | :) |
| 18:08.23 | zoa | which one ? :) |
| 18:08.29 | ManxPower | zoa, I'll be only 2 hr drive from Digium |
| 18:08.31 | stbain | fresh water o' plenty here |
| 18:08.33 | zoa | ah cool |
| 18:08.34 | cpatry | ManxPower: but u wont find chicks neither! |
| 18:08.36 | cpatry | :P |
| 18:08.42 | ManxPower | And technically it's the foothills to the mountians |
| 18:08.46 | drumkilla | zoa: heh, asteriskguru.com |
| 18:08.48 | zoa | cpatry: he could do special imports |
| 18:08.49 | fugitivo | hello |
| 18:08.50 | ManxPower | cpatry, girls are icky! |
| 18:09.32 | zoa | thanks thanks |
| 18:09.44 | YoYo | *TAP* *TAP* *TAP* hello? is this thing on? |
| 18:09.49 | drumkilla | YoYo: nope |
| 18:09.58 | ManxPower | X-Files, not for any device *I* use. |
| 18:10.03 | }cytrak{ | hehe kphone DTMF dialpad doesn't work x-lite works |
| 18:10.15 | file[laptop] | drumkilla: your sillyness level is dropping! |
| 18:10.26 | ManxPower | }cytrak{, I believe KPhone ONLY supports INBAND DTMF i.e. ULAW/ALAW |
| 18:10.54 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:11.06 | mxmasster | where can i find example table structure for the res_mysql ? |
| 18:11.29 | perd | lemonparty.org |
| 18:11.42 | perd | i'm totally banned now. |
| 18:12.00 | X-Files | ManxPower: brrr, not understand .... SIP protocol , 201 number local in ADAPTER FXO Port , 85.115.115.125 this IP ADAPTER |
| 18:12.25 | X-Files | ManxPower: maybe u can write correct dial() ? |
| 18:12.26 | stbain | }cytrak{: do a google search "kphone asterisk site:voip-info.org" |
| 18:12.33 | ManxPower | X-Files, normally you do not obtain a dialtone from the telco, you send the destination telephone number to the device and it sends the digits to the telco |
| 18:12.38 | stbain | }cytrak{: It works... just have to have the settings right |
| 18:12.50 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 18:13.00 | rculp | I have a weird fun issue |
| 18:13.02 | ManxPower | X-Files, I already gave you a correct Dial. Dial(SIP/telephonenumber@sipconfsection) |
| 18:13.07 | rculp | if anyone is willing to provide feedback |
| 18:13.12 | rculp | calls in and out work great |
| 18:13.19 | rculp | but, if I call someone and they don't answer |
| 18:13.23 | rculp | within 20000 ms |
| 18:13.26 | rculp | it hangs up |
| 18:14.03 | rculp | any idea where that setting is actually located? |
| 18:14.48 | X-Files | ManxPower: -- Executing Dial("SIP/75305101-4fab", "SIP/telephonenumber@sipconfsection") in new stack |
| 18:14.53 | X-Files | this not correct ;/ |
| 18:15.35 | X-Files | oh sorry |
| 18:15.38 | ManxPower | X-Files, replace telephonenumber with the telephone number you want to dial and replace sipconfsection with the sip.conf section for [yoursipdevice] |
| 18:15.49 | X-Files | yes yes, i how understand |
| 18:15.49 | ManxPower | Now, I can't help you further. I'm sorry. |
| 18:15.52 | X-Files | sorry :) |
| 18:16.08 | luite | rculp: RINGTIME variable in [general] section of extensions.conf ? |
| 18:16.10 | ManxPower | rculp, you may have a priority gap in extensions.conf. |
| 18:16.21 | Anthro | Does anyone know of a MacOS X softphone I could use as an extension for Asterisk? |
| 18:16.24 | ManxPower | luite, there is no standard variable called RINGTOME |
| 18:16.28 | luite | ok |
| 18:16.33 | ManxPower | or RINGTIME either. |
| 18:16.36 | luite | it was in the sample... |
| 18:16.37 | brettnem | Anthro: xlite |
| 18:16.45 | luite | but I guess it's just used as a parameter for the dial then |
| 18:16.46 | Anthro | brettnem: Free? |
| 18:16.49 | rculp | manx: I'll try that, thanks |
| 18:16.51 | ManxPower | rculp, you are not using some sissy gui interface like AMP or Asterisk@Home, are you? |
| 18:16.54 | brettnem | Anthro: yes |
| 18:17.01 | brettnem | ManxPower: hah |
| 18:17.04 | Anthro | brettnem: Keen, I'll google it. |
| 18:17.19 | brettnem | Anthro: xten.com I think |
| 18:17.20 | rculp | manx: negative |
| 18:17.22 | brettnem | xten |
| 18:17.27 | brettnem | ~xten |
| 18:17.31 | brettnem | ~xlite |
| 18:17.33 | jbot | rumour has it, xlite is at download xlite at: http://snipurl.com/5tgi | and see sample configs at http://snipurl.com/5tgj, or xlite is a free SoftPhone (software phone, requires no hardware) from xten inc, |
| 18:17.38 | brettnem | there ya go |
| 18:17.46 | rculp | manx: d-fender warned me against that and I'm holding my boss off from doing gui interface |
| 18:17.58 | ManxPower | rculp, good. The Asterisk console will be helpful to you, you can see what dialplan app was last executed before the disconnect. I still think you have a priority gap. |
| 18:18.13 | brettnem | ManxPower: hey you doing anything fancy for asterisk redundancy? |
| 18:18.30 | ManxPower | brettnem, no. We don't do that with the existing PBXs |
| 18:18.44 | brettnem | anyone doing any kind of asterisk redundancy? |
| 18:18.45 | rculp | manx: set it to 40000 and it worked great |
| 18:19.03 | [TK]D-Fender | GUI BAD!!!!! |
| 18:19.18 | brettnem | how about a TUI? |
| 18:19.23 | ManxPower | rculp, I had an issue where my SIP ATA had a 20 second timeout and would disconnect the call if it wasn't answered in 20 seconds |
| 18:19.33 | rculp | manx: n/m, that didn't fix it |
| 18:21.49 | file[laptop] | meep |
| 18:24.07 | jjones | would someone be willing to call via sip my asterisk server and see if you can hear anything? |
| 18:24.32 | brettnem | jjones: $5 i'll do it.. ;) |
| 18:24.47 | jjones | brettnem :-) it'd be worth $5 |
| 18:24.56 | Katty | jjones: sure |
| 18:26.12 | asterboy | ~docs |
| 18:26.14 | jbot | 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 |
| 18:26.21 | ManxPower | New Asterisk Motto! Asterisk: We kick Cisco's ass! |
| 18:26.58 | asterboy | Sign of a noob...nvr mind. :-> |
| 18:31.20 | zoa | who controls the bot ? |
| 18:31.28 | zoa | can i add something to the bot ? |
| 18:32.02 | cpatry | ~zoa |
| 18:32.03 | jbot | ACTION slaps zoa about a bit for cpatry |
| 18:32.14 | rculp | manx: how did you fix that issue |
| 18:32.18 | cpatry | :P |
| 18:32.25 | cpatry | jbot, zoa is a cool guy! |
| 18:32.26 | jbot | ...but zoa is already something else... |
| 18:32.27 | yxa | are there any great performance difference between using a 2.4 and 2.6 kernel with zaptel and * ? |
| 18:32.55 | cpatry | ~zoa |
| 18:32.57 | jbot | i guess zoa is a cool a cool guy. |
| 18:33.09 | X-Files | ManxPower: sorry, exten => 201,1,Dial(SIP/201@7530510) <--- correct ?? "[7530510]" in sip.conf 201 number in ADAPTER |
| 18:33.12 | zoa | ~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 |
| 18:33.13 | jbot | ...but docs is already something else... |
| 18:33.16 | InfraRed | ManxPower: too bad asterisk don't make phones ;) |
| 18:33.24 | Damin | zoa: Werd! |
| 18:33.25 | zoa | ~ docs |
| 18:33.26 | jbot | docs is, like, 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 |
| 18:33.27 | zoa | ~docs |
| 18:33.28 | jbot | methinks docs is 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 |
| 18:33.38 | zoa | hey damin |
| 18:33.44 | ManxPower | X-Files, Now, I can't help you further. I'm sorry. |
| 18:33.45 | zoa | damin im trying to get spam in the bot :) |
| 18:33.49 | zoa | it doesnt work |
| 18:33.49 | Anthro | What ports do I need to forward through my NAT router? I am setting up a * server inside the NAT and will be using BroadVoice for SIP service. |
| 18:34.01 | marcus2 | zoa, how about that channel.c jitterbuffer patch? =D |
| 18:34.05 | cpatry | ~docs |
| 18:34.07 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 18:34.10 | cpatry | there u go. |
| 18:34.12 | zoa | marcus2: 1 or 2 weeks |
| 18:34.15 | X-Files | ManxPower: only i waitin answer from you "YES or NO" |
| 18:34.18 | marcus2 | you said tuesday! |
| 18:34.19 | zoa | cpatry: its your bot ? |
| 18:34.19 | ManxPower | X-Files, Now, I can't help you further. I'm sorry. |
| 18:34.21 | zoa | yeah |
| 18:34.23 | zoa | i know |
| 18:34.23 | cpatry | nah |
| 18:34.24 | X-Files | eh |
| 18:34.25 | marcus2 | :) |
| 18:34.29 | zoa | we hit a problem with the changes in bridging |
| 18:34.30 | X-Files | bot ;/ |
| 18:34.37 | zoa | but are testing something now |
| 18:34.44 | marcus2 | bah :) |
| 18:34.46 | zoa | for stability |
| 18:34.53 | zoa | quality will be for the rest of the week |
| 18:34.59 | marcus2 | oh well, at least its not critical path |
| 18:35.03 | marcus2 | or i'd be pissed ;) |
| 18:35.09 | zoa | but its going very well |
| 18:35.15 | marcus2 | good |
| 18:35.16 | *** join/#asterisk P4C0 (i=1000@201.224.107.47) |
| 18:35.18 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 18:35.18 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 18:35.19 | zoa | it will be rock stable |
| 18:35.25 | twisted[mobile] | rocks aren't stable |
| 18:35.26 | twisted[mobile] | they roll |
| 18:35.27 | twisted[mobile] | :P |
| 18:35.32 | zoa | hundreds of millions of calls already went through it |
| 18:35.37 | P4C0 | hello, can I use a regular modem as fxo card with asterisk?? |
| 18:35.42 | zoa | before you can even touch it and spam me that it doesnt work :) |
| 18:36.03 | twisted[mobile] | Katty :) |
| 18:36.08 | Katty | twisted[mobile]: did you bring me something? |
| 18:36.11 | Katty | twisted[mobile]: like a bagel |
| 18:36.14 | marcus2 | i will break it |
| 18:36.17 | Katty | twisted[mobile]: or a cookie? |
| 18:36.30 | twisted[mobile] | katty: i woke up late and hauled ass through the casino |
| 18:36.44 | Katty | twisted[mobile]: ooh. |
| 18:36.50 | Katty | twisted[mobile]: imagine that...sleeping late |
| 18:36.56 | twisted[mobile] | yeah |
| 18:36.57 | Katty | twisted[mobile]: now /why/ would you be doing that? |
| 18:36.59 | stbain | P4C0: not really |
| 18:37.00 | twisted[mobile] | after no sleep in almost 48 hours |
| 18:37.23 | Katty | twisted[mobile]: have you made your peace with starbucks yet? |
| 18:37.29 | twisted[mobile] | i couldn't today |
| 18:37.32 | Katty | :< |
| 18:37.34 | twisted[mobile] | the line was out to the casino |
| 18:37.42 | Katty | ouch. |
| 18:37.46 | twisted[mobile] | *nods* |
| 18:37.51 | twisted[mobile] | but the showers here rock :) |
| 18:38.07 | Katty | do they have caffinated showers? |
| 18:38.19 | P4C0 | :D |
| 18:39.41 | zoa | did you bring me something ? |
| 18:39.50 | Katty | twisted[mobile]: there's also this weird tapping noise around my desk. |
| 18:39.58 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 18:39.59 | zoa | GHOSTS! |
| 18:40.01 | [TK]D-Fender | Hey, what are the odds on getting app_queue to use dialplan HINTS with AgentCallBackLogin? |
| 18:40.04 | zoa | i see dead people |
| 18:40.16 | Katty | i see dead pixels |
| 18:40.32 | file | i see dead inodes |
| 18:40.39 | Katty | maybe there's a mousey |
| 18:41.23 | Damin | Katty: The question that I will soon answer (later this afternoon) is wether a call originating as G.729 from the CCM and being handed back to the CCM w/ a G.729 speaking phone will work properly if ASterisk is just speaking Ulaw in the middle.. |
| 18:41.49 | Wunsch | Can anybody point me at some useful documentation for the h323 channel in Asterisk? Specifically, I'd like to connect Asterisk to an Avaya phone system that speaks H.323, not just have H.323 clients on my Asterisk. |
| 18:41.58 | tmccrary | that won't work in pass-thru mode |
| 18:42.05 | tmccrary | I think you'll need a license for the codec |
| 18:42.20 | tmccrary | then asterisk can transcode the audio |
| 18:42.21 | Katty | Damin: yay? |
| 18:42.31 | Katty | Damin: you're clearly missing the bigger picture. |
| 18:43.14 | Damin | Katty: Probably.. ;) |
| 18:45.20 | kuku5 | I have a problem with * when a user records a greeting - but that greeting doesnt play when someone calls in ( voicemail ) |
| 18:45.33 | *** join/#asterisk jtodd (n=jtodd@204.96.162.40) |
| 18:46.03 | Katty | kuku5: are you sure they're recording the right greeting? |
| 18:46.43 | Katty | kuku5: there's only 5 of them, afterall. |
| 18:48.00 | [TK]D-Fender | 5 messages? |
| 18:48.18 | [TK]D-Fender | I only recall 3 (busy, unavail, temp), and your name (which doesn't really count |
| 18:48.28 | *** join/#asterisk bweschke (n=bweschke@204.96.162.40) |
| 18:51.52 | *** join/#asterisk Corydon76-lap (n=corydon@pdpc/supporter/sustaining/Corydon76-home) |
| 18:52.02 | kuku5 | hm |
| 18:52.05 | ManxPower | Why does "your name" not count? |
| 18:52.13 | kuku5 | where is "your name" show up |
| 18:52.22 | *** join/#asterisk ken___ (n=ken@host-66-59-246-138.lcinet.net) |
| 18:52.26 | ken___ | anyone around ? |
| 18:52.30 | ManxPower | kuku5, in the Directory App |
| 18:52.46 | ken___ | i'm trying to get automon (options wW on Dial command) to work correctly |
| 18:52.55 | ken___ | but i have no idea how to do it, etc. |
| 18:52.59 | ken___ | i have features.conf setup correctly |
| 18:53.01 | [TK]D-Fender | it also shows up in the "canned" messages if you didn't record a custom message. |
| 18:53.36 | ManxPower | ken___, nothing turned up during your extensive searches of the Wiki and the mailinglist archives? |
| 18:53.54 | ken___ | ManxPower: no, nothing came up |
| 18:54.14 | ManxPower | ken___, it's a fairly new feature |
| 18:54.22 | ken___ | ManxPower: the information that did come up said to just enable it in features.conf and then enable the wW on the Dial command |
| 18:54.30 | ken___ | which, i've done |
| 18:54.36 | X-Files | ManxPower: http://pastebin.ca/28828 --> i turn sip debug, in line 635 - 673 send DTMF correct ? i dunno this correct |
| 18:54.38 | ken___ | but, i have no idea where the recordings would be going |
| 18:54.45 | ken___ | nor do i see any kind of feedback on the CLI |
| 18:54.46 | ManxPower | ken___, I think you must have sox installed if you want to mix the call sides |
| 18:54.54 | kuku5 | Playing 'vm-intro' (language 'en') |
| 18:54.57 | kuku5 | This is what it plays |
| 18:55.02 | ken___ | ok, great -- i have sox installed! |
| 18:55.08 | ken___ | so how do i get it to work ? any idea ? |
| 18:55.14 | kuku5 | Katty: Playing 'vm-intro' (language 'en') << why does it play this |
| 18:55.30 | ManxPower | ken___, your best bet is to ask on the mailinglists |
| 18:55.57 | Katty | kuku5: because that's the intro of voicemail |
| 18:56.06 | ken___ | ManxPower: ugh -- i dread doing that... there's too much noise on the asterisk-users list |
| 18:56.15 | kuku5 | katty: but it should play the person's greeting |
| 18:56.17 | ManxPower | ken___, We all have our crosses to bear. |
| 18:56.49 | ManxPower | X-Files, Now, I can't help you further. I'm sorry. |
| 18:57.29 | ManxPower | ken___, "show application mixmonitor" that might work better for you |
| 18:57.31 | brc_ | anybody got a number for ELI's ops center? trying to get a pri turned up |
| 18:57.35 | X-Files | ManxPower: brrr... you are ignored me ? |
| 18:58.25 | ManxPower | X-Files, Now, I can't help you further. I'm sorry. |
| 18:58.32 | [TK]D-Fender | kuku5 : what you need to do is like "exten => 2,2,Voicemail(su1234)". s = don't add the canned instructions, u=unavailable message (use "b" for busy message where you feel like it...) |
| 18:58.45 | kuku5 | Katty: it should ahh |
| 18:58.48 | kuku5 | so wee ned the S |
| 18:58.57 | [TK]D-Fender | kuku5 : the reason for the intro is the "s" you had |
| 18:58.57 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
| 18:59.01 | [TK]D-Fender | or didn't rather... |
| 18:59.04 | X-Files | ken___: please say to ManxPower this line : http://pastebin.ca/28828 --> i turn sip debug, in line 635 - 673 send DTMF correct ? i dunno this correct |
| 18:59.04 | ManxPower | kuku5, "show application voicemail" will tell you the options you can use |
| 18:59.05 | [TK]D-Fender | correct |
| 18:59.10 | [TK]D-Fender | WIKI!!!! |
| 18:59.12 | [TK]D-Fender | ~wiki |
| 18:59.20 | kuku5 | Katty: on previous versions i didnt ahve to do that |
| 18:59.46 | Katty | kuku5: which greeting did you record? |
| 19:00.11 | Katty | kuku5: it's playing vm-intro because it didn't see anything valid to play in front of it |
| 19:00.13 | }cytrak{ | hey guys is this correct ? AGI (/opt/asteris/.../agi-bin/myperl | ${train}) |
| 19:00.16 | kuku5 | hm |
| 19:01.38 | *** part/#asterisk _m_ (n=m@nat-ph3-1.rz.uni-karlsruhe.de) |
| 19:01.48 | *** join/#asterisk _m_ (n=m@nat-ph3-1.rz.uni-karlsruhe.de) |
| 19:02.42 | kuku5 | - Executing VoiceMail("Zap/2-1", "s69") in new stack |
| 19:02.42 | kuku5 | Nov 15 14:01:54 DEBUG[21993]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals |
| 19:02.42 | kuku5 | <PROTECTED> |
| 19:02.53 | kuku5 | i dont think its the s |
| 19:03.35 | yxa | sorry my english is not so good. i have been trying to find out what "IAX provisioning" means. be great if someone gives me a primer... |
| 19:03.48 | Katty | kuku5: when you call voicemail... |
| 19:03.55 | Katty | kuku5: and it says press 5 or whatever for advanced options |
| 19:04.01 | Katty | kuku5: or maybe it's 3 for greeting......anyway |
| 19:04.09 | Katty | kuku5: which greeting /precisely/ are you recording? |
| 19:04.21 | kuku5 | s = skip message |
| 19:05.01 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 19:05.03 | kink0 | re |
| 19:05.08 | Katty | rehi |
| 19:05.16 | Katty | kuku5: s is not a valid option. |
| 19:05.20 | kink0 | I got my first license for g729 ;)ç |
| 19:05.20 | Katty | kuku5: which greeting are you recording? |
| 19:05.21 | kuku5 | yes it is |
| 19:05.32 | Katty | i give up |
| 19:05.54 | kink0 | well , now I need at least one more license, to try with the other point ussing g.729 |
| 19:06.01 | kink0 | I hope not burn cpu !! |
| 19:06.08 | yxa | is provisioning only use for iaxy devices? |
| 19:06.15 | Katty | i need at least one more soda. |
| 19:06.22 | ManxPower | Katty, "show application voicemail" says "s" is a valid option./ |
| 19:06.29 | Katty | ManxPower: i'm not talking about that. |
| 19:06.34 | ManxPower | Katty, OK |
| 19:06.35 | Katty | ManxPower: i'm talking about when you dial in to get your voicemail. |
| 19:06.46 | Katty | ManxPower: press 3 or whatever to record greetings |
| 19:06.49 | Katty | ManxPower: etc. |
| 19:06.52 | ManxPower | Katty, Ah, well that's "voicemailmain" |
| 19:07.05 | ManxPower | Katty, it's "0" to get to the options |
| 19:07.16 | Katty | ManxPower: if he didn't record the right greeting.. |
| 19:07.22 | Katty | ManxPower: then it's not going to play any greeting |
| 19:07.30 | ManxPower | Katty, yup |
| 19:07.59 | Katty | right. |
| 19:08.36 | *** join/#asterisk wolfson (n=ggggg@208.25.254.120) |
| 19:09.02 | kuku5 | how do i turn this off: 15 14:08:27 DEBUG[21993]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample inter |
| 19:09.14 | InfraRed | Katty: christmas is in *FUCKEN DECEMBER* |
| 19:09.27 | X-Files | ManxPower: http://pastebin.ca/28828 --> i turn sip debug, in line 635 - 673 send DTMF correct ? i dunno this correct |
| 19:09.31 | InfraRed | it's a day |
| 19:09.33 | ManxPower | kuku5, you don't. It's a harmless debug message. If you don't want to see it, don't start asterisk with a -d |
| 19:09.38 | InfraRed | not 2 whole months |
| 19:09.45 | kuku5 | i didnt start it with a d |
| 19:09.48 | kuku5 | just with vv's |
| 19:09.58 | ManxPower | kuku5, then you need to look at logger.conf |
| 19:09.59 | Katty | InfraRed: ... |
| 19:10.00 | Katty | InfraRed: gosh |
| 19:10.02 | kink0 | Call accepted by 216.207.245.8 (format gsm) <-- digium does not support g729 or must I set up g729 as preferible codec ? |
| 19:10.05 | Katty | InfraRed: simmer down. |
| 19:10.27 | *** join/#asterisk saftsack (n=saftsack@p54A7C949.dip.t-dialin.net) |
| 19:10.27 | ManxPower | kink0, If you want G729 then you must buy a G729 license from the patent holder via Digium |
| 19:10.40 | shmaltz | ManxPower, you found a colo? |
| 19:10.44 | kink0 | ManxPower, yes, I bougth now, and I have registered |
| 19:10.47 | ManxPower | shmaltz, not really |
| 19:10.54 | kink0 | Found total of 1 G.729 licenses |
| 19:10.58 | InfraRed | kink0: g729 is ok without license in passthrough mode |
| 19:11.02 | ManxPower | kiko69, If ANY other codec is allowed, Asterisk will pick that codec instead of G729 |
| 19:11.07 | ManxPower | shmaltz, not really |
| 19:11.26 | ManxPower | It's amazing how many responses I got from colos from New York to Dallas |
| 19:11.26 | kink0 | ManxPower, ahh ok, even if I set in order first allow to g729 ? |
| 19:11.27 | shmaltz | any prefference on where it's located? |
| 19:11.33 | kuku5 | How do I change the verbosity |
| 19:11.36 | saftsack | guest (Unspecified) D 255.255.255.255 0 Unmonitored |
| 19:11.38 | kuku5 | under the CLI |
| 19:11.39 | ManxPower | kink0, no, you need to disallow=all and allow=g729 |
| 19:11.48 | saftsack | is this correct for accepting any voip telephones? |
| 19:11.54 | ManxPower | kuku5, set verbose blah |
| 19:11.57 | ManxPower | and set debug off |
| 19:12.06 | ManxPower | but since debug is not verbose they are different things |
| 19:12.10 | kink0 | ManxPower, yes , but ONLY g729 for everybody or would be enough to set g729 as the first one ? |
| 19:12.28 | ManxPower | kiko69, If ANY other codec is allowed, Asterisk will pick that codec instead of G729 |
| 19:12.59 | kink0 | ManxPower, then if I set g729 there no way my PBX works with the rest of people who has not g729 , right ? |
| 19:13.08 | ManxPower | kink0, so if you want to force everyone to only use g729 then in [general] disallow=all allow-g729 If you want to only force G729 for SOME devices, then you would put those lines in the device section of sip.conf |
| 19:13.20 | ManxPower | kink0, I don't know. I don't manage your PBX |
| 19:13.46 | kink0 | well, really I don't manage it yet !!! but I am learning :) |
| 19:14.10 | ManxPower | kink0, how do you conenct to your PBX? |
| 19:14.16 | ManxPower | T-1/E-1/SIP/H323? |
| 19:14.31 | kink0 | ManxPower, worst !! just a soundcard and CLI now |
| 19:14.45 | ManxPower | kink0, just kill yourself now and save all the pain you have in the future. |
| 19:15.16 | kink0 | ManxPower, xDDDDDDDDDDD I have take a bit of masoquist before start to learn Asterisk !! |
| 19:15.38 | zoa | kink0: we did that when there was no documentation at all |
| 19:15.40 | zoa | nothing |
| 19:15.44 | zoa | just try and see what happens |
| 19:16.01 | zoa | even the show application thing was not doing what it should do |
| 19:16.15 | kink0 | zoa: I am lucky then, here is the docs |
| 19:16.23 | zoa | hehe |
| 19:16.24 | zoa | yeah |
| 19:16.26 | zoa | many of them |
| 19:16.31 | zoa | and many more coming every day |
| 19:16.43 | kink0 | but proof&fails is needle |
| 19:16.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:17.00 | kink0 | yes, too much doc, for a simple idea |
| 19:17.13 | ManxPower | kink0, Step 1: Don't be a cheap ass, but phones and ATAs and interface cards. |
| 19:17.14 | *** join/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net) |
| 19:17.21 | oej | That was the first complain I ever seen in Asterisk that we have TOO MUCH DOCUMENTATION! |
| 19:17.29 | mog_work | lol |
| 19:17.33 | mog_work | who said that oej |
| 19:17.46 | ManxPower | oej, There is a lot of documentation, it's just all disorganized and mostly wrong. |
| 19:17.47 | zoa | yeah |
| 19:17.51 | zoa | its not wrong |
| 19:17.53 | ManxPower | kink0, read the new Asterisk book |
| 19:17.54 | kink0 | ManxPower, the scenary is I pretend to terminate to GSM, so I saw how try to connect a GSM terminal to a sound card. |
| 19:17.56 | oej | (20:16:39) kink0: yes, too much doc, for a simple idea |
| 19:17.56 | ManxPower | ~docs |
| 19:17.57 | jbot | i heard 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 |
| 19:17.57 | zoa | things are getting better eveyr day |
| 19:18.02 | zoa | it was wrong before |
| 19:18.05 | zoa | but its getting better |
| 19:18.11 | kink0 | I am not cheap, but Stargate racks are a bit expansive just to start. |
| 19:18.13 | ManxPower | kink0, GSM CELL is not the same as VoIP GSM |
| 19:18.27 | ManxPower | kink0, you use WHATEVER iterface the GSM terminal has. |
| 19:18.33 | oej | I am a bit worried about voip-info.org quality |
| 19:18.35 | X-Files | ManxPower: http://pastebin.ca/28828 --> i turn sip debug, in line 635 - 673 send DTMF correct ? i dunno this correct |
| 19:18.36 | ManxPower | Usually a E-1 |
| 19:18.44 | kink0 | ManxPower, sure, I know, what is the diference between network and codec |
| 19:18.46 | ManxPower | oej, only a little bit? *grin* |
| 19:19.05 | zoa | i think things will grow ok in the end |
| 19:19.14 | zoa | once asterisk changes less |
| 19:19.18 | zoa | now its just too much changes |
| 19:19.22 | kink0 | ManxPower, yes, there an option to buy Teles,2N,Valiant ... or so, with or without voIP card, with only E1 |
| 19:19.22 | zoa | but i think thats almost over |
| 19:19.43 | oej | zoa: Asterisk won't change less |
| 19:19.47 | kink0 | ManxPower, are you in Europe ? ( I deduced because you speak about E1 instead T1 ) |
| 19:19.48 | ManxPower | kink0, I don't know GSM terminals except for what I read on the Asterisk mailinglist |
| 19:20.06 | oej | ManxPower: You do need to move over here... |
| 19:20.17 | ManxPower | kink0, no, I'm in the USA, but NOBODY in the USA/Canada uses GSM terminals, so I assumed you must not be in USA/Canada |
| 19:20.34 | ManxPower | oej, too late. Nobody offered me a job so I bought a car and am moving to the mountians |
| 19:20.37 | kink0 | I know I will need to buy some Teles/2N or so GSM gateways, but I read about Asterisk and I hope will be a perfect complement |
| 19:20.52 | Beirdo | head for the hills! |
| 19:21.01 | oej | ManxPower: Far away from the coast and the storms? |
| 19:21.03 | ManxPower | kink0, it would be much easier to connect the GSM terminal directly to your PBX |
| 19:21.09 | ManxPower | oej, YES!!!! |
| 19:21.16 | ManxPower | specifically about 2 hrs from Digium |
| 19:21.34 | saftsack | my voip telephone isnt recongnized by asterisk but i can ping it |
| 19:21.35 | ManxPower | oej, I'm told there are waterfalls on the property that I am looking at. |
| 19:21.43 | zoa | no shit |
| 19:21.48 | Beirdo | ManxPower: nice |
| 19:21.50 | zoa | thats wet ManxPower |
| 19:21.58 | ManxPower | I'll see the property this coming weekend. |
| 19:22.02 | kink0 | ManxPower, in USA for mobile termination calls are originated from proper lines ? |
| 19:22.05 | zoa | cool take pictures |
| 19:22.08 | oej | ManxPower: Where is that located? |
| 19:22.20 | kink0 | here, is more expansive proper->mobile than mobile->mobile |
| 19:22.23 | ManxPower | kink0, In the USA mobile carriers do not permit you to connect directly to the carrier. |
| 19:22.56 | ManxPower | oej, 100km NE of Birmingham AL, 300km SE of Huntsville AL (Digium) |
| 19:22.56 | *** join/#asterisk loick (n=loick@APuteaux-151-1-61-134.w82-120.abo.wanadoo.fr) |
| 19:23.04 | zoa | -> going home (=> means taking the elevator one floor up) |
| 19:23.25 | zoa | 300km, at american driving speeds thats a day, not 2 hours :op |
| 19:23.27 | zoa | :p |
| 19:23.30 | oej | ManxPower: Sounds like a good location |
| 19:23.33 | ManxPower | zoa, Hush you |
| 19:23.38 | kink0 | ManxPower, well, here are not very kindly when you pretend to connect to mobile |
| 19:23.44 | InfraRed | Recruitment agencies only seem to exist to make used car salesmen, |
| 19:23.44 | ManxPower | oej, I have not made a final decision yet. |
| 19:23.44 | InfraRed | lawyers, and estate agents look ethical. |
| 19:23.46 | Nugget | it's definitely 2 hours of alabama driving speed. |
| 19:23.47 | InfraRed | LOL |
| 19:23.55 | zoa | ManxPower: good luck, send us pictures!!! |
| 19:24.05 | Katty | ooh, a Nugget |
| 19:24.06 | zoa | and invite us all for a big bbq |
| 19:24.09 | zoa | i will take the beer |
| 19:24.10 | zoa | :) |
| 19:24.13 | saftsack | no one want to help me? :( |
| 19:24.21 | zoa | add bear repellant too :) |
| 19:24.29 | ManxPower | zoa, In the usa, freeway, you can assume about 1 mile per minute |
| 19:24.39 | Katty | saftsack: be patient and ask again in 10 minutes |
| 19:24.43 | saftsack | ko |
| 19:24.44 | saftsack | ok |
| 19:24.46 | *** join/#asterisk slak- (i=slak@rewted.biz) |
| 19:24.54 | Katty | Nugget: i demand hug. |
| 19:25.00 | slak- | hey i got asterisk-stat-v2 workin, i'd like to add a Cost column to it |
| 19:25.06 | slak- | anyone have that going? |
| 19:25.11 | Katty | k, all better. |
| 19:25.36 | kink0 | saftsack, nmap your IP phone |
| 19:25.43 | slak- | also, im using 1.2.0 beta-1, whats the urgency of upgrading? |
| 19:25.51 | slak- | should i wait til 1.2 final? |
| 19:26.21 | Beirdo | Nugget: that'd be pretty close to Lynchburg, TN then, no? |
| 19:27.42 | kink0 | Call rejected by 216.207.245.8: Unable to negotiate codec |
| 19:28.01 | kink0 | appears not to work if i set : disallow=all ; allow=g729 |
| 19:28.18 | saftsack | kink0, i deinstalled asterisk and now i follo the 10min instruction |
| 19:29.34 | slak- | whats the best asterisk gui manager for windows |
| 19:29.38 | slak- | similar to gastmanm |
| 19:30.04 | Nugget | Beirdo: km, not miles. |
| 19:30.25 | Beirdo | oops, misread :) |
| 19:30.26 | Beirdo | heh |
| 19:30.41 | Beirdo | my brain was thinking american for some reason |
| 19:30.42 | kink0 | may be digium does not support g729 at all ? |
| 19:30.58 | file | on misery? probably not |
| 19:31.14 | slak- | file: suggest an asterisk gui manager application |
| 19:31.16 | zoa | slak-: switchvox i think |
| 19:31.43 | zoa | watch out with file, before you know it he will be sending you prank msn messages in the middle of the night :) |
| 19:31.59 | file | will not! |
| 19:32.02 | zoa | just pretend you didnt see him :) |
| 19:32.08 | file | and yeah, I like switchvox - their software is cool, and so are the people who make it |
| 19:32.19 | zoa | he's just a little jitter on the irc line |
| 19:32.27 | zoa | yeah she's nice :) |
| 19:32.32 | slak- | switchvox isnt free |
| 19:32.33 | file | but married. |
| 19:32.35 | slak- | ;/ |
| 19:32.35 | zoa | and she still speaks to me, its amazing |
| 19:32.41 | file | you want everything for free? too bad :) |
| 19:32.51 | zoa | file, georgi would say: you can't feel that |
| 19:33.01 | file | :P |
| 19:33.05 | kink0 | Call rejected by 216.207.245.8: Unable to negotiate codec :( |
| 19:33.05 | slak- | i dont know..whats the point of things like gastman |
| 19:33.11 | slak- | isnt the CLI good enouhg? |
| 19:33.24 | file | kink0: we got it the first time you pasted that |
| 19:34.53 | zoa | try selling that cli to a customer |
| 19:35.18 | zoa | haha, changing the dialplan through the cli, would somebody ever do that ? |
| 19:35.23 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 19:35.29 | zoa | besides the freaky admin here in the company ? |
| 19:35.41 | saftsack | Nov 16 21:41:29 WARNING[16770]: chan_mgcp.c:4050 reload_config: Unable to get our IP address, MGCP disabled |
| 19:35.58 | slak- | okay i need a windows manager that actually works |
| 19:35.58 | slak- | heh |
| 19:36.01 | zoa | thats a good thing! |
| 19:36.13 | funxion | can anyone tell me exactly what ${CHANNEL} looks like from a zap channelk? |
| 19:37.20 | saftsack | anyone knows help for me? |
| 19:37.50 | funxion | would Zap/1-1 be zap group 1 channel 1 or would the output be different? |
| 19:38.53 | ManxPower | funxion, Zap/1-1 would be the Zap channel 1, the first call. |
| 19:39.09 | ManxPower | Zap/1-2 would be first zap channel, 2nd call (call waiting, three-way calling, etc) |
| 19:40.11 | funxion | ManxPower Im trying to manipulate the channel number into a variable but I dont currently hav a box to test with that has a zap card in it |
| 19:40.46 | ManxPower | fugitivo, channel formats are TECH/DEVICE-CALLID |
| 19:40.50 | funxion | if I was to call ${CHANNEL} and the call came in on zap group 1 channel 1 what would the output of the channel variable be |
| 19:41.33 | *** join/#asterisk Mad_Hornet (n=trodasta@70-254-74-98.ded.swbell.net) |
| 19:42.44 | saftsack | what are reasons for that asterisk isnt able to determine my ip adress? |
| 19:42.48 | ManxPower | So you could do something like Cut(TECHNOLOGY=CHANNEL,/,1) |
| 19:43.00 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-966c9ed333730cc4) |
| 19:43.06 | *** part/#asterisk Mad_Hornet (n=trodasta@70-254-74-98.ded.swbell.net) |
| 19:44.52 | funxion | ManxPower would that only give me the group number? |
| 19:45.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 19:48.55 | ManxPower | funxion, um, you never see the group number in ${CHANNEL} |
| 19:48.59 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
| 19:49.23 | *** join/#asterisk Kort (n=james@65.211.216.202) |
| 19:49.24 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 19:49.24 | funxion | o |
| 19:49.28 | funxion | didnt know that |
| 19:49.33 | Kort | anyone know why asterisk might skip a priority 1 rule? |
| 19:49.33 | funxion | its just th channel number |
| 19:49.37 | gambolputty | What billing software do any of you typically use? |
| 19:49.41 | Kort | and execute the priority 2 rule? |
| 19:50.01 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 19:50.04 | funxion | then kewl thnx ManxPower |
| 19:50.50 | [TK]D-Fender | Kort, shouldn't happen naturally, maybe on a GOTO that was done wrong. pastebin a sample w/ extensions.conf |
| 19:51.06 | Kort | there are no goto's its just an include |
| 19:51.08 | zoa | gambolputty: our own |
| 19:51.10 | Kort | in a context. |
| 19:51.21 | Kort | exten => _91NXXNXXXXXX,1,Dial(${TRUNK}${EXTEN:${TRUNKMSD}}) ; Standard long distance |
| 19:51.24 | Kort | exten => _91NXXNXXXXXX,2,Dial(${BACKUP_TRUNK}${EXTEN:${TRUNKMSD}}) |
| 19:51.28 | Kort | first one never gets executed. |
| 19:51.28 | gambolputty | zoa: Is yours open source or commercial? |
| 19:51.34 | Kort | only the 2nd. |
| 19:51.37 | zoa | its none of both |
| 19:51.47 | Kort | however if you copy the first one to the 2nd priority, then it works |
| 19:51.49 | zoa | its internal code |
| 19:51.58 | gambolputty | that's what I thought |
| 19:52.12 | zoa | there were plans to sell it, but i'm not sure that will actually ever happen |
| 19:52.31 | zoa | its overkill for normal people |
| 19:52.47 | gambolputty | but meets your own needs |
| 19:53.09 | zoa | yeah, its billing for big masses |
| 19:53.13 | gambolputty | ok |
| 19:53.22 | zoa | so you need a trained person to touch it |
| 19:54.03 | zoa | the good thing is, it can do a lot of things, the bad thing is, it can do a lot of things :) |
| 19:54.17 | marcus2 | http://www.krisk.org/tinypbx/pics/ |
| 19:54.17 | marcus2 | nice |
| 19:55.33 | zoa | On Tue, 2005-11-15 at 10:17 -0800, Will Glass-Husain wrote: |
| 19:55.34 | zoa | >> On Windows, I really like TextPad (shareware - www.textpad.com ) for editing |
| 19:55.34 | zoa | >> text files. |
| 19:55.34 | zoa | I like edlin cause its old school :P |
| 19:55.34 | zoa | -- |
| 19:55.36 | zoa | looooooool |
| 19:55.36 | zoa | :) |
| 19:56.26 | [TK]D-Fender | Kort : pastebin your extensions.conf please... I''d like a better look at the big picture including your variables.... |
| 19:56.53 | Kort | actually I just found the problem, hehm |
| 19:56.58 | [TK]D-Fender | EDLIN : for that vintage DOS 2.2 feel! |
| 19:56.59 | [TK]D-Fender | whee! |
| 19:57.09 | [TK]D-Fender | <- OLD |
| 19:57.15 | InfraRed | edlin \o/ |
| 19:57.42 | InfraRed | it's where emacs went wrong |
| 19:57.43 | InfraRed | :D |
| 19:58.04 | [TK]D-Fender | Actually I think the oldest version I used was 2.1. Before that was CP/M and a few others... |
| 19:58.12 | InfraRed | but seriously for windows i'd use ultraedit |
| 19:58.22 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:58.27 | InfraRed | i started with dos 3.3 |
| 19:58.34 | InfraRed | then down to 2.1 :P |
| 19:58.53 | saftsack | i can ping my voip telephone but asterisk doesnt get it |
| 19:59.06 | InfraRed | saftsack: throw it away |
| 20:00.15 | saftsack | lol |
| 20:00.23 | *** join/#asterisk lidl (n=little@213-140-22-71.fastres.net) |
| 20:00.32 | lidl | i'd like to ask if there's a plan for electrical outages in places where asterisk is used |
| 20:00.38 | lidl | i mean, using the normal phone, even if there's an outage, the phone still works |
| 20:01.47 | saftsack | InfraRed, aah now there is a message |
| 20:01.57 | lidl | but if every call goes through *, then no electricity -> no calls |
| 20:02.38 | InfraRed | lidl: same if you use a normal PBX |
| 20:02.43 | InfraRed | it'll go dead |
| 20:02.50 | InfraRed | unless you get UPS |
| 20:03.40 | lidl | is there kinda of a mechanical 'switch' to re-route calls from asterisk to a bunch of phone? |
| 20:04.06 | sahafeez | lidl: i have a ups on my box and get 40 mins of uptime |
| 20:04.20 | InfraRed | lidl: a UPS |
| 20:04.35 | InfraRed | only issue will be with POE for the phones |
| 20:04.43 | skyen | My pbx's got about on month worth of diesel backing it's racksize ups ;) |
| 20:04.48 | skyen | one* |
| 20:05.09 | lidl | i see |
| 20:05.17 | InfraRed | your rack runs on diesel? |
| 20:05.17 | InfraRed | :) |
| 20:05.28 | *** join/#asterisk enemy^x (i=lkqw@212.62.250.98) |
| 20:05.36 | lidl | uhm, unleaded fuel ;) |
| 20:05.50 | enemy^x | cdr_mysql.conf (is it possible to configure a remote mysql server?) Tried changing localhost to a remote one, but it still complains about the sock file. |
| 20:06.01 | skyen | InfraRed: no, but the generators backing the ups does |
| 20:06.01 | ^Howler | no, the lead helps rebuild the firewall.. |
| 20:06.09 | skyen | haha |
| 20:07.25 | *** part/#asterisk Anthro (n=dsfgrt@pdpc/supporter/active/Anthro) |
| 20:07.33 | JonR800 | enemy^x: it complains but it works. |
| 20:07.41 | lidl | http://www.voip-info.org/wiki/view/PSTN+Pass-through |
| 20:07.42 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 20:07.52 | docelmo | Hay *NIX guys what IRC client do you use? |
| 20:07.53 | lidl | a pass-through is a good solution to me |
| 20:08.19 | InfraRed | lidl: yes but you will need analogue only phones |
| 20:08.25 | InfraRed | wont work with ip phones |
| 20:08.25 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 20:08.41 | InfraRed | i have pass through on my FXS box |
| 20:08.46 | InfraRed | works nicely |
| 20:08.47 | lidl | ok, i'll have some old phone just-in-case |
| 20:08.51 | ^Howler | docelmo: I'm using irssi right now.. |
| 20:08.53 | InfraRed | look at mediatrix kit |
| 20:09.09 | tmccrary | gaim |
| 20:09.14 | docelmo | I just converted to Linux/Wine Hybrid.. |
| 20:09.33 | InfraRed | docelmo: sick man |
| 20:09.40 | ^Howler | docelmo: mIRC =) |
| 20:09.41 | LostFrog | I am having a horrible problem with a snom360. It keeps asking for the password even though it is provisioned and registered.. |
| 20:09.42 | azzie | docelmo, we are using XML irc client on Cisco 79xx phones |
| 20:09.52 | docelmo | I cant get 100% away from windows.. work.. Not my choice.. |
| 20:10.01 | loud | i use epic. |
| 20:10.01 | LostFrog | I mean, it works for 5 minutes, then it stops working. |
| 20:10.08 | tmccrary | i don't know what a linux/wine hybrid is, but it sounds cool |
| 20:10.17 | tmccrary | I dropped a snom phone before, 1 1/2 feet.... it died |
| 20:10.18 | enemy^x | jonR800: Yes, I just noticed that it actually works. The strange thing is that it complains (Nov 15 21:07:54 WARNING[19983]: cdr_addon_mysql.c:330 my_load_module: MySQL database sock file not specified. Using default)..... That's strange?! Since it shouldnt care about the sock file when using the port? right? |
| 20:10.27 | docelmo | Its Linux that runs Windows application |
| 20:10.27 | docelmo | s |
| 20:11.02 | JonR800 | enemy^x: right :) im sure it's just a logic issue, but no need to worry, it works. |
| 20:12.43 | paryl | in order to log agents out i set up this extension based on examples: exten => 2002,1,Dial(Local/2003/n,,D(#)) ...but i get "No such extension/context 2003@default creating local channel" and " Unable to create channel of type 'Local'" |
| 20:12.43 | docelmo | For instance.. I am running Winamp 5 and Linux |
| 20:12.43 | ^Howler | loud: do you like epic? |
| 20:13.02 | [TK]D-Fender | paryl : you need a context for that dial cmd... Dial(Local/2003@somecontexthere) |
| 20:13.04 | paryl | oh sigh, i realized my error after typing the question... nevermind |
| 20:13.11 | paryl | thanks fender :) |
| 20:13.37 | [TK]D-Fender | np |
| 20:14.44 | kink0 | oppppsssssssss great surprise !! I do a call, and hear the answering machine in the other side, ussing a modem |
| 20:14.53 | paryl | one related question though... i have 2 basically identical asterisk installs, both are 1.0.9, compiled from source, but one doesn't have the 'dial' command at the console |
| 20:14.54 | *** join/#asterisk gniretar (i=mark@152.160.35.157) |
| 20:15.01 | gniretar | hi people |
| 20:15.05 | kink0 | but I have not connected any audio between modem ( external unit ) and my sound card !! |
| 20:15.30 | kink0 | how is sound from the PSTN call going to the sound card ? |
| 20:15.45 | kink0 | the RS-232 ??? |
| 20:16.05 | hardwire | yup |
| 20:16.29 | hardwire | 8000hz conversations fit into 64k very well |
| 20:16.50 | hardwire | your system can speak to a modems controller at 115k if it wants |
| 20:17.06 | kink0 | I can not believe ... since I have not any audio connected between external modem ( is just a rs232 cable ) |
| 20:17.34 | kink0 | hardwire, ussing the DAC on the modem and setting it for voice, and then ussing data on the serial ? |
| 20:17.37 | hardwire | well its not magic |
| 20:18.04 | kink0 | hardwire, yeah, but really I was expecting to need to connect the audio ports on the modem to the audio ports on the sound card |
| 20:18.17 | hardwire | you could |
| 20:18.20 | gniretar | can anyone help me with this issue? |
| 20:18.21 | gniretar | http://pastebin.com/430789 |
| 20:18.29 | gniretar | it is on a new SuSe 10.0 installation |
| 20:19.04 | hardwire | kink0: I just popped in |
| 20:19.08 | hardwire | what are you doing and how? |
| 20:19.28 | kink0 | hardwire, I have just a soundcard and external modem |
| 20:19.38 | hardwire | but an fxo pci :) |
| 20:19.41 | hardwire | but/buy |
| 20:19.43 | kink0 | then I dialed ( ussing modem ) a PSTN number |
| 20:19.45 | hardwire | kink0: but I like your style. |
| 20:19.48 | gniretar | it is a te210P card |
| 20:20.04 | kink0 | hardwire, I pretend to go up to 60 GSM channels !! hehehee |
| 20:20.07 | JonR800 | gniretar: i don't know the answer.. but how did you install asterisk? |
| 20:20.24 | gniretar | JonR800: from source |
| 20:20.42 | gniretar | as well as the zap drivers |
| 20:20.51 | gniretar | from source |
| 20:20.54 | JonR800 | gniretar: i see.. and you did libpri / zaptel with a make install? |
| 20:20.57 | kink0 | hardwire, yes, I have here at home some fxo and fxo, but are not digium, they are quicknet pci cards |
| 20:21.08 | gniretar | JonR800: yes |
| 20:21.38 | gniretar | i have done this before. never with this perticular card though |
| 20:22.14 | docelmo | So whadup? |
| 20:22.35 | docelmo | Is anyone here buying domestic for less than .0092 w/ no commit? |
| 20:22.40 | JonR800 | k, i was hoping it was just a simple noob mistake :) (might still be, im rather noobish myself) |
| 20:23.54 | hardwire | kink0: ah |
| 20:24.51 | ^Howler | docelmo: prefer imported, myself |
| 20:25.29 | *** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net) |
| 20:25.39 | gniretar | this must be a noobish mistake |
| 20:25.43 | gniretar | hmm |
| 20:25.46 | gniretar | i hate SuSe |
| 20:25.50 | gniretar | with a passion |
| 20:25.56 | gniretar | i bet its the funk kernel they use |
| 20:25.59 | gniretar | funky* |
| 20:26.02 | *** join/#asterisk Leob (n=chatzill@70.20.25.221) |
| 20:27.01 | Leob | Hello there, what's the best way to convert from MP3 to wav? |
| 20:28.31 | morale | mpg123 |
| 20:28.51 | kink0 | time to dinner !! cu later ! |
| 20:28.53 | morale | mpg123 -w filename.wav song.mp3 |
| 20:28.55 | *** join/#asterisk Tclp (n=Tcalp@S0106000c4191c793.ed.shawcable.net) |
| 20:29.21 | Leob | would that be enough to produce files to be played by Asterisk? |
| 20:30.24 | *** join/#asterisk bweschke_ (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 20:30.27 | gniretar | Leob when do you wanna play them? |
| 20:30.41 | Tclp | hey all, I have a fairly unrelated question -- we have a Vonage line here at the office and I'm hoping to be able to drop into this line via some sort of PC-to-PC call (Home to Office to the VOIP Line), I'm not sure what type of hardware/software would be required for something like this ... we do have some old unused Asterisk boxes that are no longer being used bu I'm hoping to not have ot goto that extent |
| 20:30.45 | *** join/#asterisk rubicant (n=rubicant@ut-n-35192.adsl.wanadoo.nl) |
| 20:30.50 | rubicant | hey, they say your penis gets small when you buy a xbox360 |
| 20:31.10 | gniretar | rubicant: /topic |
| 20:31.23 | Leob | I want to play them for my caller during the call |
| 20:31.28 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net) |
| 20:32.08 | docelmo | Imported shit.. beer.. geesh |
| 20:32.13 | gniretar | music on hold? |
| 20:32.20 | *** part/#asterisk rubicant (n=rubicant@ut-n-35192.adsl.wanadoo.nl) |
| 20:33.09 | Leob | nope |
| 20:33.34 | gniretar | is it music? |
| 20:33.41 | gniretar | or voice |
| 20:33.48 | Leob | yes, mainly music |
| 20:34.00 | gniretar | u wan this to play during the convo? |
| 20:34.15 | bzbw | Hi, got a issue with DISA, in "exten => 9,1,DISA,no-password|Dial_Outside", how do I set up Dial_Outside context with pattern mapping instead defining each extension number? |
| 20:34.25 | Leob | ? |
| 20:34.44 | gniretar | describe to me exactly what you wanna do |
| 20:35.19 | Katty | take a nap. |
| 20:35.27 | Leob | I'm implementing an audioblog tool that reads mp3 files from certain sites and plays them for the user on his/her phone |
| 20:35.56 | gniretar | hmm |
| 20:35.57 | gniretar | well |
| 20:36.04 | gniretar | it is possible |
| 20:36.15 | gniretar | do do this and keep the mp3 format |
| 20:36.19 | gniretar | not sure how |
| 20:36.24 | gniretar | part of aserisk addons i think |
| 20:36.48 | Leob | I'm not too concerned about keeping the mp3 format |
| 20:37.05 | gniretar | mp3 is a good format |
| 20:37.09 | gniretar | keep it |
| 20:37.11 | gniretar | lol |
| 20:37.24 | gniretar | unless you can convert it to gsm |
| 20:37.27 | gniretar | dunno how though |
| 20:37.57 | Leob | ok... I'll keep trying... |
| 20:39.10 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 20:40.06 | bzbw | emm, maybe I should put my question this way: Anyone know that I can define context with pattern like _9xxxx? |
| 20:40.40 | skyen | rephrase yourself, please. |
| 20:41.42 | bzbw | I thought I made it clear: Can we define context with pattern? i.e., 6xxx where x is any number from 0-9. |
| 20:41.54 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 20:42.12 | wunderkin | i doubt it |
| 20:42.15 | [TK]D-Fender | Well its not a CONTEXT you are defining, but an EXTEN inside of one, and yes you can. |
| 20:42.43 | [TK]D-Fender | exten => 6xxx,1,DoSomethingNifty() |
| 20:42.50 | [TK]D-Fender | umm |
| 20:42.54 | [TK]D-Fender | exten => _6xxx,1,DoSomethingNifty() |
| 20:42.57 | [TK]D-Fender | better :) |
| 20:43.33 | bzbw | TK: not sure understand you, I'm trying to use exten => 9,1,DISA,no-password|My_Pattern_Context |
| 20:44.06 | IronHelix | well you could do exten => _6xxx,1,Goto(context,${EXTEN},1) |
| 20:44.07 | bzbw | where My_Pattern_Context is the context that I want to define for North America call |
| 20:44.09 | IronHelix | or use Fork |
| 20:44.59 | X-Files | have users used "Internet Telephony Gateway" ????? |
| 20:45.03 | bzbw | This is to allow someone to call my pbx then calling out to any NorthAmerica number |
| 20:45.35 | IronHelix | yeah then you want DISA |
| 20:46.43 | IronHelix | use disa to put them in the context that lets them call a north america dial pattern |
| 20:46.52 | bzbw | IronHelix: I tried _6xxx or other pattern mapping, it is not working when used in DISA. |
| 20:47.35 | IronHelix | well if you do _6xxx,1,DISA it will give you disa for any 4digit exten that starts with 6 |
| 20:47.55 | IronHelix | how bout this |
| 20:48.00 | IronHelix | tell me what exactly you want to dial |
| 20:48.03 | IronHelix | and what should happen |
| 20:48.10 | IronHelix | and i'll tell you want you need to script |
| 20:49.50 | bzbw | I want for someone to dial into my pbx, then exten => 9,1,DISA,passcode|Dial_Outside_PSTN, which allow someone to use this PBX to dial any PSTN number Defined in Dial_Outside_PSTN |
| 20:49.55 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 20:50.39 | IronHelix | that will work, but i'd format it as exten => 9,1,DISA(password,Dial_Outside_PSTN) |
| 20:50.45 | IronHelix | also keep in mind contexts are case sensitive |
| 20:51.54 | *** join/#asterisk Timoti (n=asqsa@85.99.166.94) |
| 20:52.06 | bzbw | that's fine, the issue is there are SO many PSTN numbers, I don't know what the user will call, so I can not define all of them in Dial_Outside_PSTN, I have to use Pattern mapping |
| 20:52.13 | Timoti | are any of you using H323 with asterisk |
| 20:53.12 | Timoti | noone ? |
| 20:53.14 | IronHelix | ok so in your [Dial_Outside_PSTN] you can put something like exten => _1XXNXXXXXX,1,Dial(SIP/yourprovider/${EXTEN}) |
| 20:53.46 | bzbw | great, thanks IronHelix, that's what I need. |
| 20:53.50 | IronHelix | that will allow them to dial any 11 digit number and it will go out on the SIP account that starts with [yourprovider] |
| 20:54.31 | Timoti | ironhelix .. do you have exprience with H323 with asterisk ? |
| 20:54.34 | *** join/#asterisk darby_t (i=darby_t@dku26.neoplus.adsl.tpnet.pl) |
| 20:54.45 | IronHelix | sadly no, i only use SIP and IAX |
| 20:55.00 | Timoti | :-( |
| 20:55.07 | IronHelix | whats the problem tho |
| 20:55.10 | IronHelix | maybe i can figure it out |
| 20:55.24 | IronHelix | if you want to take the risk of typing it out possibly for nothing that is |
| 20:55.28 | Timoti | I have a 48 port fxs with H323 .. I would like to use it with astersik |
| 20:55.33 | Timoti | I have seen this one |
| 20:55.45 | brettnem | Hey, anyone know why a "sip show peers" would suddenly cause many peers to become UNREACHABLE? this happens alot |
| 20:55.51 | Timoti | http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/ |
| 20:55.59 | Timoti | want to install on my asterisk |
| 20:56.19 | IronHelix | i dont think you need to install openH323 anymore, as i recall asterisk comes with its own chan_h323 |
| 20:56.28 | Timoti | but I have not that much exprience about where ( or to which directory ) I should install it |
| 20:56.46 | Timoti | really |
| 20:56.51 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
| 20:56.53 | christo | evening all |
| 20:57.12 | IronHelix | hi |
| 20:57.59 | hhoffman | hi |
| 20:58.28 | IronHelix | well tim, generally you'd download it to /usr/src |
| 20:58.35 | IronHelix | then tar -zxvf the file |
| 20:58.39 | IronHelix | cd to the new directory |
| 20:58.41 | IronHelix | and do make |
| 20:58.43 | IronHelix | then make install |
| 20:58.49 | IronHelix | do that before you compile asterisk i think |
| 20:58.54 | IronHelix | unless you need to patch asterisk itself |
| 20:59.32 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-79.cpe.netcabo.pt) |
| 21:00.04 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
| 21:00.30 | IronHelix | http://www.voip-info.org/wiki/index.php?page=Asterisk+H323+channels this might have some useful info |
| 21:01.12 | IronHelix | either wya, the package should have some kindof readme or install instructions included with the download |
| 21:02.06 | }cytrak{ | not sure if anyone using perl-agai but anyways ... I'm trying to pass $gsm = "digits/200" to $AGI->stream_file($gsm_file, $digits) bu ti keep getting file does not exist |
| 21:02.32 | *** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 21:02.32 | }cytrak{ | I think asterisk thinks the whole file name is "digits/200" |
| 21:03.47 | shido6 | you do need openh323 and pwlib |
| 21:03.49 | shido6 | for chan_h323 |
| 21:06.14 | *** join/#asterisk IOscanner (n=IOscanne@216-165-210-74.crescentb.com) |
| 21:06.22 | }cytrak{ | is there a way can set the root of sound files within an AGI ? |
| 21:07.01 | IOscanner | I get WARNING: Symbol version dump /usr/src/linux-2.6.12.3/Module.symvers |
| 21:07.01 | IOscanner | <PROTECTED> |
| 21:07.02 | IOscanner | <PROTECTED> |
| 21:07.20 | IOscanner | I have kernel source that I build the kernel from and a few other kernels for different processors |
| 21:07.58 | IOscanner | I build asterisk against this version a while back but now I can't build against the same kernel 2.6 source. |
| 21:08.02 | IOscanner | any ideas? |
| 21:09.31 | IOscanner | dmesg has this: zaptel: version magic '2.6.12.3 preempt 486 gcc-3.3' should be '2.6.12.3 preempt CYRIXIII gcc-3.3' Looks like the kernel source is not intune with what I am running |
| 21:09.38 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:09.50 | IOscanner | Can I cross compile for the new proccessor |
| 21:10.24 | justinu | how would I troublehoot major lag in meetme... using ztdummy |
| 21:11.28 | IronHelix | ioscanner- is /usr/src/linux-2.6 symlinked to the correct version of your kernel source? |
| 21:11.39 | IOscanner | Yes |
| 21:12.03 | IOscanner | 2.6.12.3 |
| 21:13.14 | X-Files | People that perfect SIP or H.323? |
| 21:13.31 | IronHelix | perfect? |
| 21:13.41 | IronHelix | i dont know that any protocol is 'perfect'... |
| 21:13.47 | X-Files | :) |
| 21:13.58 | InfraRed | SIP sucks |
| 21:14.00 | X-Files | People, that perfect protocol SIP or H.323? |
| 21:14.03 | InfraRed | use morse code |
| 21:14.07 | IronHelix | lol |
| 21:14.17 | azzie | IronHelix, IP over dove network was pretty good |
| 21:14.18 | azzie | :) |
| 21:14.23 | IronHelix | lol |
| 21:14.40 | InfraRed | azzie: get it right |
| 21:14.43 | InfraRed | avian carrriers |
| 21:15.03 | InfraRed | http://www.faqs.org/rfcs/rfc2549.html |
| 21:15.24 | X-Files | more more comments, please |
| 21:15.26 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:16.00 | IronHelix | xfiles- are you asking if you should use SIP or 323? |
| 21:16.07 | InfraRed | X-Files: use SIP |
| 21:16.10 | IronHelix | becuase if so, i say use SIP |
| 21:16.17 | InfraRed | unless your providers only uses H323 |
| 21:16.24 | InfraRed | SIp is not very nat friendly btw |
| 21:16.35 | IronHelix | 323 is worse |
| 21:16.40 | IronHelix | at least SIP has STUN |
| 21:16.40 | X-Files | ;) |
| 21:16.46 | X-Files | IronHelix: yep :) |
| 21:17.38 | IronHelix | SIP is also much easier to get working with * |
| 21:17.42 | IronHelix | ie it works out of box |
| 21:17.42 | X-Files | i try configure ITG protocol SIP and asterisk 5 day, and can't get normal work .. |
| 21:18.18 | InfraRed | asterisk 5? |
| 21:18.23 | X-Files | no :) |
| 21:18.28 | IronHelix | ITG? what exactly are you trying to do? connect IP phones to asterisk, connect asterisk to provider, or connect asterisk to asterisk? |
| 21:18.32 | X-Files | i configure 5 day ;( |
| 21:18.35 | IronHelix | he has been trying for 5 days |
| 21:18.39 | IronHelix | and it isnt working yet |
| 21:18.41 | IronHelix | i think |
| 21:18.42 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 21:18.56 | X-Files | ITG = Internet Telephony Gateway |
| 21:19.10 | InfraRed | asterisk sucks , use microsoft telephone server |
| 21:19.31 | X-Files | microsoft sucks too :) |
| 21:19.40 | IronHelix | xfiles- what is the make and model of your itg? |
| 21:20.07 | X-Files | eusso utg7104 |
| 21:21.05 | X-Files | 2 FXO and 2 FXS ports :) |
| 21:21.29 | IronHelix | and what does not work? does the gateway not register to asterisk, do calls not go through? |
| 21:21.52 | X-Files | big problem, i can't send to FXS port from ASTERISK "number tone" |
| 21:22.25 | X-Files | IronHelix: all registred to asterisk.. |
| 21:22.58 | IronHelix | DTMF is broken |
| 21:22.59 | IronHelix | hmmm |
| 21:23.09 | X-Files | yep |
| 21:23.14 | IronHelix | crab, brb |
| 21:23.19 | IronHelix | *crap |
| 21:23.39 | *** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com) |
| 21:23.41 | X-Files | yes this crap |
| 21:24.52 | X-Files | read rfc2833 |
| 21:25.15 | }cytrak{ | anyone knows how I can pass a path to stream_file ? |
| 21:25.32 | X-Files | no helping |
| 21:25.36 | InfraRed | in asterisk.conf |
| 21:25.46 | InfraRed | }cytrak{: ^ |
| 21:26.49 | }cytrak{ | that's not what i mean |
| 21:27.03 | X-Files | http://pastebin.ca/28828 |
| 21:27.04 | X-Files | line 635 - 673 send DTMF |
| 21:27.40 | *** join/#asterisk liran_ (n=liran@80.178.87.126.adsl.012.net.il) |
| 21:28.11 | liran_ | i've got some basic questions about asterisk Master.csv log file |
| 21:28.21 | liran_ | could anyone be of help please? |
| 21:28.25 | X-Files | IronHelix: have comments ? |
| 21:28.33 | Dr_Ray | liran - what is your question? |
| 21:29.06 | liran_ | Dr_Ray, Thanks, firstly, are all the calls made by all yours concentrated in that file? |
| 21:29.22 | Dr_Ray | yes, unless you set it up to split it |
| 21:30.17 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 21:30.19 | liran_ | Dr_Ray, ok |
| 21:30.42 | liran_ | Dr_Ray, i've noticed some special keywords there like MEETME, what does it mean? |
| 21:31.11 | BleedingMe | anyone come across a good solution for a door phone that will only dial the receptionist... like just a call button. We need something that can be mounted outside so the receptionist can buzz people in. |
| 21:31.22 | Dr_Ray | meetme is the application |
| 21:31.39 | Dr_Ray | payphone.com sells a phone like that |
| 21:31.58 | shido6 | is $200 a decent price for used cisco 7960s? |
| 21:32.00 | shido6 | I have 2 of them |
| 21:32.01 | Dr_Ray | under no dial |
| 21:32.15 | Dr_Ray | I'd pay $200 for a 7960 |
| 21:32.18 | liran_ | Dr_Ray, well i mean there are keywords like echo playback and meetme, what does each mean? |
| 21:32.19 | BleedingMe | cool.. thanks Dr_Ray |
| 21:32.21 | X-Files | Ppls: i set to sip.conf : dtmfmode=info , paste debug ->>> http://pastebin.ca/28843 Please check |
| 21:32.53 | shido6 | I have 2 for sale - |
| 21:32.55 | *** join/#asterisk bweschke (n=bweschke@204.96.162.40) |
| 21:33.03 | shido6 | with firmware and ringtones |
| 21:33.10 | shido6 | & powercube |
| 21:33.12 | Dr_Ray | bleeding - then you just set it to immediate mode |
| 21:34.25 | clyrrad | Ok so I have call forwarding as *72 in the dial plan and its being included in the phones context. Also on the CLI I can see the DBPut command running. However when I call the phone that I just tried to forward it rings normally. What am I doing wrong? |
| 21:35.17 | kink0 | off-topic somebody knows how enhanced sound quality from a voice modem ? ( yeahh I am ussing Dial,Modem ) |
| 21:35.55 | liran_ | are there perl/php tools that knows how to parse the Master.csv logfile and output a nice report of rates and calls? |
| 21:35.56 | Dr_Ray | liran - asterisk breaks down stuff into commands, dial meetme, playback, record.. that is what you are seeing |
| 21:36.14 | liran_ | thanks Dr_Ray |
| 21:36.20 | Dr_Ray | liran - I wrote a perl script to process my calls |
| 21:36.22 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 21:36.24 | liran_ | i'm not looking for a full-scale billing solution |
| 21:36.42 | liran_ | Dr_Ray, is it something you can send over? |
| 21:36.51 | Dr_Ray | not really, it's maybe 20 lines |
| 21:36.59 | Tclp | hey all, I have a fairly unrelated question -- we have a Vonage line here at the office and I'm hoping to be able to drop into this line via some sort of PC-to-PC call (Home to Office to the VOIP Line), I'm not sure what type of hardware/software would be required for something like this ... we do have some old unused Asterisk boxes that are no longer being used bu I'm hoping to not have ot goto that extent |
| 21:37.23 | liran_ | Dr_Ray, is it calculating rates also? |
| 21:38.10 | Dr_Ray | liran - it just charges 10 cents per minute for calls |
| 21:38.41 | liran_ | right but these needs to be calculated. like say the total amount of time was 10 minutes, then it should report 10cents*10 |
| 21:39.23 | Dr_Ray | right, perl can do that easily |
| 21:39.42 | *** join/#asterisk nuba (i=nuba@zaxxon.telerama.com) |
| 21:39.45 | liran_ | well i'd rather use something that's already doing it well than re-invent the wheel |
| 21:40.04 | Dr_Ray | it took me 4 hours to reinvent that wheel, it won't take you that long |
| 21:40.08 | liran_ | i dont suppose you can direct me to some links of free open-source tools like that? |
| 21:40.15 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 21:40.33 | liran_ | i dont know how much is it going to take me... i dont want to do it if it's already been done :) |
| 21:41.00 | Dr_Ray | well, good luck |
| 21:41.45 | rezEdit | Can anyone provide some guidance on how I would set up a 'hunt group' - an extension that when called tries any number of extensions in a particular order before going to voicemail? I know I can write my own dialplan logic for this but I am thinking this has to be a pretty common thing for people to want to do... perhaps there is some application that provides it. |
| 21:43.29 | shido6 | you *could* use queues |
| 21:43.42 | shido6 | but a macro or some dialplan logic would be easier |
| 21:44.32 | rezEdit | shido6: right, ok. thanks. I will have a look at queues |
| 21:45.02 | Netgeeks | I've got something that I just can't seem to get a grip on here: http://pastebin.ca/28844 |
| 21:45.27 | Netgeeks | Seems the app Directory is pissed about something and I just can't seem to figure out what. Anyone have any thoughts? |
| 21:46.07 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 21:46.37 | rezEdit | Netgeeks: are you using the Directory app in your extension.conf? |
| 21:46.54 | rezEdit | er extensions.conf |
| 21:46.56 | *** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il) |
| 21:47.05 | Netgeeks | yep, you can see from the show dialplan main-menu in pastebin |
| 21:47.12 | rezEdit | right, duh |
| 21:47.42 | IronHelix | rezedit- how about a multiple dial string |
| 21:48.02 | IronHelix | Dial(SIP/phone1&SIP/phone2&SIP/phone3) |
| 21:48.11 | IronHelix | it will ring them all simultaneously tho |
| 21:48.40 | rezEdit | IronHelix: riiight :-) I will bee needing that for something else, but it won't work for what I am doing now. |
| 21:49.28 | X-Files | bljaaaa |
| 21:49.30 | X-Files | ;( |
| 21:49.41 | *** part/#asterisk SplasPood (i=nobody@paravolve.net) |
| 21:49.41 | rezEdit | IronHelix: I actually have a wonky AGI script that I found online that is supposed to find all registered SIP phones and call them simultaneously. |
| 21:49.58 | IronHelix | wow |
| 21:50.10 | rezEdit | IronHelix: It's a bit broken though so I will probably do it all manually until I can build my own. |
| 21:50.17 | IronHelix | hehe |
| 21:50.43 | IronHelix | xfiles- is your dtmfmode set to what the gateway is expecting? |
| 21:53.05 | asterboy | Asterisk, "The Future of Telephony", Price at Oreilly.com $57, Price a Amazon.ca $37 hmmmmmm....which one to buy? |
| 21:53.21 | IronHelix | or download it for free |
| 21:53.27 | IronHelix | if you can deal with reading on the screen |
| 21:53.41 | asterboy | can't read it on the throne then. |
| 21:54.38 | IronHelix | lol |
| 21:55.09 | Katty | or you can just bribe one of your friends who can get it for free |
| 21:55.09 | rezEdit | asterboy: no laptop? :-) |
| 21:55.26 | Katty | and then turn it into a scrapbook and have all your developy friends sign you a little message on it |
| 21:55.34 | asterboy | lol...a little bulky though. |
| 21:56.00 | rezEdit | asterboy: and the right laptop would keep you toasty warm, too! |
| 21:56.06 | *** join/#asterisk Corydon76-lap (n=corydon@pdpc/supporter/sustaining/Corydon76-home) |
| 21:56.20 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:56.44 | Netgeeks | Hello Katty, sorry, I'm out of soy milk today |
| 21:57.09 | Netgeeks | Anyone else want to take a glance at http://pastebin.ca/28844 and see if you can figure out where I'm messed up? |
| 21:57.12 | asterboy | Katty, ya thats a great idea, I had Gerard Beekmans, founding father of Linux From Scratch, sign my book when I bought him lunch. |
| 21:58.54 | asterboy | I'd like to do that with Jim and Jared |
| 21:59.10 | Netgeeks | I wonder if anyone ever got John Postel to sign a printed email from netsol for an early IP allocation... that would be neat |
| 21:59.19 | rezEdit | Netgeeks: it's looking in the 'extensions' context but your mailbox is in 'default'? |
| 22:00.02 | Katty | asterboy: (= |
| 22:00.24 | Katty | Netgeeks: s'ok, i'm busy plotting chocolate chip peanut butter pancakes dusted in confectioner's sugar |
| 22:00.37 | Netgeeks | rez: nah, it's getting the voicemail context right, if you see in the top section, it's actually pulling the right info out of voicemail.conf 'Directory(default|extensions) where default is the voicemail context and extensions is the extension.conf context for the dial-context |
| 22:01.00 | Netgeeks | I've even given it a extensions.conf extension with _X. to match anything and it still errors |
| 22:01.31 | devel | isn't enumlookup() supposed to jump to priority n+101 if it doesn't find a record (or jump out of the context if there is no n+101)? |
| 22:02.15 | exstatica | anyone had trouble with a polycom 301? |
| 22:02.40 | cnet2 | I saw the polycom 301 has half duplex speakerphone >:SSS |
| 22:02.40 | exstatica | i created a config file and it works fine on my 501, but my 301 tries to use the username "default" and no password |
| 22:02.56 | asterboy | wow $21 at bookpool.com |
| 22:03.40 | cnet2 | i've seen posts of people having trouble with polycom 501 and asterisk.. anyone with this? I'm think'n of recommending this phone to my company |
| 22:03.50 | justinu | they work fine |
| 22:03.53 | exstatica | my 501 works great with asterisk |
| 22:04.00 | devel | cnet2, i agree, they work fine |
| 22:04.27 | rezEdit | cnet2: I am setting up about 30 polycom 501's now, no probs so far. |
| 22:04.42 | exstatica | if you use a config file make sure you configure the mwi |
| 22:04.47 | shido6 | 7960s (2) for 200 ea |
| 22:04.54 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 22:04.54 | cnet2 | great.. :D |
| 22:05.18 | exstatica | anyone ever setup a 301 on asterisk? |
| 22:05.21 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 22:05.43 | Beirdo | argh |
| 22:05.50 | justinu | shouldn't really be much different than a 501 |
| 22:05.51 | justinu | or 601 |
| 22:06.04 | exstatica | that's what i thought but for some reason it's not obeying the username/password |
| 22:06.04 | Beirdo | where's JerJer when I need to ask him something? :) |
| 22:06.07 | rezEdit | Netgeeks: Sorry, a bit of a newbie here and I can't see anything that might be causing the error you're seeing there. |
| 22:06.21 | devel | exstatica, yeah, we have 301 too |
| 22:06.33 | exstatica | do you use a tftp config? |
| 22:06.41 | Netgeeks | yeah, rez, I think I might have found a bugged Directory app. telling the admin of that system to upgrade his version |
| 22:06.43 | justinu | exstatica: which username/password? |
| 22:06.52 | asterboy | Anyone want to sell S100M module? |
| 22:06.57 | asterboy | FXS |
| 22:07.02 | devel | we haven't yet, exstatica. i'm behind in my dev schedule. |
| 22:07.08 | test34 | http://www.asterisk.org/changelog only list the changes in asterisk-1.2.0-beta1... where can I see changelog from rc1 to rc2 ? |
| 22:07.09 | rezEdit | exstatica: Don't those use a slightly different config file? I seem to remmeber seeing something about that. Are you reusing your 501 config format for the 301's? |
| 22:07.11 | exstatica | reg.1.auth.userId= |
| 22:07.18 | exstatica | yeah i am |
| 22:07.25 | Beirdo | ~seen JerJer |
| 22:07.32 | jbot | jerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 4d 14h 10m 1s ago, saying: 'not really a debian specific question, but someone here should know - Can i merge partitions in Linux? like my / was created way too small and i would like to blow away another partiton and start over, but one issue is I am currently not ... |
| 22:07.35 | exstatica | that's what i wanted to know, but i havn't been able to find a 301 config |
| 22:07.36 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 22:07.38 | justinu | which sip version does it run? |
| 22:07.48 | Beirdo | wow, he's been gone for a while |
| 22:07.56 | rezEdit | exstatica: Do you have the SoundPoint IP Admin Guide? |
| 22:08.09 | exstatica | not sure, it did an upgrade of firmware like my 501's do |
| 22:08.22 | exstatica | rezEdit: maybe somewhere |
| 22:09.45 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 22:10.26 | rezEdit | exstatica: my bad, looks like it should all be the same from what I can see. |
| 22:11.01 | rezEdit | exstatica: if you hard code it on the phone does it work? |
| 22:11.58 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 22:13.49 | justinu | i've had issues with 501s not respecting those parameters myself |
| 22:13.54 | skyen | How can i make asterisk present me a dialtone so that a can make it place a call for me? |
| 22:14.05 | *** join/#asterisk kn0x (n=nunya@tor/session/x-56665d7f0e2469e4) |
| 22:14.25 | justinu | i went into the admin menu and said "reset local config" |
| 22:14.28 | skyen | I'm building a dialback-service so that i can make a phonecall to my pbx, and make it dial me back and present me a dialtone |
| 22:14.36 | file | skyen: DISA |
| 22:14.42 | skyen | stands for? |
| 22:14.51 | justinu | ~disa |
| 22:14.52 | jbot | i heard disa is direct inward system access. show application disa |
| 22:15.27 | exstatica | i havnt' tried |
| 22:15.56 | asterboy | Interesting that the clone X100 cards only need to have the R13 and R19 connects removed with a soldering iron to turn them into genuine cards. |
| 22:15.57 | exstatica | i'll boot it up right now and check |
| 22:17.28 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 22:17.28 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-156-65.44-151.net24.it) |
| 22:17.35 | asterboy | guess its just a Intell v92 winmodem card. |
| 22:17.53 | asterboy | no wonder the TDM400s are not cloned yet.. |
| 22:18.05 | *** join/#asterisk santiago (n=santiago@63.245.87.62) |
| 22:18.12 | tzanger | asterboy: no, that's the x101p |
| 22:18.12 | file | <cough>they are</cough> |
| 22:18.19 | tzanger | the TDM400 is not a winmodem |
| 22:18.22 | X-Files | Hey PPL, Need help, i call to FXO port and redirect to ASTERISK. Asterisk answer and ask phone nubmer, i enter in numpad my phone "203" , DTFM working |
| 22:18.23 | tzanger | and they're already cloned |
| 22:18.31 | justinu | i don't understand why people use pci cards instead of analog sip gateways.... price? |
| 22:18.36 | kn0x | can someone runnign 2.6.13 gentoo kernel show me there kernel config |
| 22:18.41 | kn0x | i cant get ztdummy runnign |
| 22:18.56 | brad_mssw | asterboy: eh, are you saying that you _had_ to make a modification to the X100P clone to make it work as an FXO board? |
| 22:18.57 | kn0x | ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control |
| 22:19.07 | asterboy | sip gates are not cheap, no? |
| 22:19.17 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 22:19.18 | kn0x | us what dmesg shows after modprobe ztdummy |
| 22:19.18 | justinu | i guess not |
| 22:19.27 | tzanger | justinu: because external boxes are a tangle of wires and wall warts |
| 22:19.38 | asterboy | brad_mssw: no, its something I read...I think the ones on ebay already have the modification so they can tote them as OEM genuine. |
| 22:19.47 | kn0x | is anyone running 2.6.13 w/ ztdummy i should say? |
| 22:19.57 | X-Files | Try : call from FXS port to ASTERISK. Asterisk answer and ask phone nubmer, i enter 201 and i connecting to line FXO port, there dialtone and me need dial nubmer , my dial numbers ignored |
| 22:20.10 | justinu | tzanger: but those digium cards generate all sorts of interrupts and stuff... ;) |
| 22:20.21 | asterboy | tzanger: are the TDMs cloned??? |
| 22:20.28 | X-Files | DTFM not worked from asterisk to FXO port .. |
| 22:20.32 | asterboy | tzanger: Can't find them anywhere. |
| 22:20.34 | X-Files | why ? |
| 22:20.55 | tzanger | justinu: so? |
| 22:21.03 | tzanger | asterboy: don't worry about it |
| 22:21.07 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 22:21.09 | kn0x | nobody running 2.6.13? |
| 22:21.10 | tzanger | they aren't significantly cheaper |
| 22:21.27 | asterboy | lol, so it is out there...but you have to dig...ok. |
| 22:21.47 | kn0x | zaptel needs rtc right? |
| 22:21.57 | X-Files | pplz please help |
| 22:22.00 | kn0x | does Generic RTC 1.0.7 good enough? |
| 22:22.20 | IronHelix | xfiles is dtmfmode set correctly in sip.conf? |
| 22:22.34 | asterboy | ASterix categorized as Christian book: http://www.biblestudynotes.org/cgi-bin/estore/onlinestore.cgi?item_id=0596009623&search_type=AsinSearch&templates=1&locale=us |
| 22:22.45 | kn0x | he said from fxs to fxo i believe ironhelix |
| 22:22.59 | kn0x | i dont understand what hes asking really to be honest |
| 22:23.25 | IronHelix | i was talking to him earlier |
| 22:23.37 | IronHelix | he has a strange voice gateway |
| 22:23.43 | IronHelix | and his dtmf is not working |
| 22:23.45 | X-Files | IronHelix: i trying dtmfmode=rfc2833 , not work |
| 22:23.46 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 22:23.52 | kn0x | oh i see |
| 22:23.59 | X-Files | IronHelix: and dtmfmode=inband not work |
| 22:24.05 | IronHelix | did you try =info? |
| 22:24.09 | X-Files | IronHelix: and dtmfmode=info to same |
| 22:24.33 | X-Files | but info i see Sip read: |
| 22:24.33 | X-Files | SIP/2.0 501 Not Implemented |
| 22:24.49 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 22:24.56 | X-Files | IronHelix: http://pastebin.ca/28843 line 62 debug in dtmfmode=info |
| 22:25.43 | InfraRed | X-Files: try inline |
| 22:25.55 | file[laptop] | inline isn't even valid ;) |
| 22:26.04 | file[laptop] | only valid options are rfc2833, inband, and info |
| 22:26.08 | *** join/#asterisk bartpbx (n=bartpbx@p54B0360E.dip0.t-ipconnect.de) |
| 22:26.12 | bartpbx | hello |
| 22:26.17 | X-Files | ;) |
| 22:27.06 | kn0x | who runs RTC as a module for zaptel? |
| 22:27.06 | X-Files | what me doing, how ? |
| 22:27.16 | InfraRed | dtmfmode=rfc2833 |
| 22:27.18 | InfraRed | ignore me |
| 22:27.19 | InfraRed | :L) |
| 22:27.23 | InfraRed | hometime |
| 22:27.24 | InfraRed | \o |
| 22:27.40 | asterboy | Found something: Dialogic Cards http://www.tti.net/computer-telephony/dialogic.html |
| 22:28.04 | justinu | ugh, dialogic |
| 22:28.21 | justinu | don't remind me of those times |
| 22:28.33 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 22:28.40 | X-Files | ehhh |
| 22:29.02 | bartpbx | I have little question. I found information in the wikki on "# Dial returns ${CAUSECODE}: If the dial failed, this is the errormessage" But this does not seam to work in HEAD. what is the "new" way to get the reason for a dial failure? |
| 22:29.36 | X-Files | InfraRed: have comments ? |
| 22:29.44 | drumkilla | bartpbx: ${DIALSTATUS} |
| 22:31.13 | bartpbx | drumkilla, $DIALSTATUS only returns "NOANSWER" but i need to know why |
| 22:31.29 | bartpbx | e.g. no route to host, timed out, circuit busy.... |
| 22:32.27 | drumkilla | HANGUPCAUSE, then |
| 22:32.28 | drumkilla | :) |
| 22:32.57 | bartpbx | ah |
| 22:34.18 | QbY | WHY?! Are these doing this.. In my queue_log -- Nothing is being recorded.. It shows the call entering the queue, being answered and being completed in the same second--the calls are lasting in excess of 10-15 minutes.. 1132093363|1132093336.79|310|NONE|ENTERQUEUE||"Barrington IL" <x> |
| 22:34.18 | QbY | 1132093395|1132093336.79|310|Local/9x@from-internal-7580,1|CONNECT|32 |
| 22:34.18 | QbY | 1132093395|1132093336.79|310|Local/9x@from-internal-7580,1|COMPLETECALLER|32|0 |
| 22:34.30 | rayvd | Thx for that. |
| 22:34.42 | bartpbx | ah.. hangupcause is good |
| 22:34.52 | drumkilla | sweet :) |
| 22:35.49 | file[laptop] | I just got another job offer... |
| 22:35.52 | X-Files | drumkilla: You do not wish to help me, I have " Internet Telephony Gateway " from it I can to call on asterisk and naberat numbers from ports FXS and FXO, but the problem that asterisk cannot transfer dtmf in port FXS and FXO when there is a long hooter |
| 22:35.58 | marcus2 | i wonder where i can find an asterisk consultant that actually has experience with other pre-voip PBX systems |
| 22:36.12 | X-Files | drumkilla: naberat = enter |
| 22:41.01 | kn0x | does anyone know anything about the rtc patches for ztdummy in 2.6 ??!?!?!?! |
| 22:41.18 | kn0x | im trying to add rtc_register and some others but im stuck |
| 22:41.51 | Beirdo | naptime |
| 22:43.21 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:46.18 | oogle | are there any resources on how channel numbers are generated for zaptel cards? |
| 22:46.32 | IronHelix | based on pci slots and available channels |
| 22:46.36 | oogle | because if i put a t1 card in a machine, and two TDM24xx cards... how can i know which channel |
| 22:46.48 | IronHelix | depends on which order they are in |
| 22:46.50 | oogle | will be assigned to each card without trial and error |
| 22:47.02 | IronHelix | like if you put the tdm card in first, the t1 will start with ch5 i think |
| 22:47.36 | oogle | but what if i put them in at the same time? |
| 22:47.42 | IronHelix | i mean in the first pci slot |
| 22:47.43 | oogle | is it what order they appear in lspci? |
| 22:47.50 | IronHelix | yeah i think so |
| 22:47.52 | oogle | ok |
| 22:48.57 | oogle | thanks for the pointer IronHelix |
| 22:49.02 | IronHelix | np |
| 22:49.08 | IronHelix | also if your tdm card has empty slots |
| 22:49.10 | IronHelix | they still count |
| 22:49.45 | IronHelix | so like if you have a tdm card with one module, and then a t1 card after it, the t1 will start at ch5, channels 2-4 will be empty but reserved |
| 22:50.13 | oogle | oh thanks a lot, i didn't know that |
| 22:50.29 | oogle | because they only give partially filled cards to customers |
| 22:50.34 | oogle | heh |
| 22:50.42 | oogle | (my employer) |
| 22:50.46 | IronHelix | hehe |
| 22:50.47 | marcus2 | just use gentoo, it makes it easy to add the rtc patches to ztdummy ;) |
| 22:51.16 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 22:51.16 | oogle | marcus2: as much of a fan I am of Gentoo (I use it at home), I wouldn't use it for production |
| 22:51.17 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:51.33 | marcus2 | well of course not, its linux |
| 22:52.01 | oogle | linux is totally production ready, after all 4/5 of the fastest computers in the world run it |
| 22:52.14 | oogle | and if those aren't production machines, i don't know what is |
| 22:52.31 | marcus2 | heh |
| 22:52.54 | mog_work | i think people that say linux is not production ready are just a ignorant or b willfully ignorant.... |
| 22:52.54 | marcus2 | what distros are those systems running? |
| 22:53.12 | oogle | mog_work: or on Microsoft's payroll |
| 22:53.14 | justinu | oracle supports linux |
| 22:53.20 | mog_work | that is b |
| 22:53.22 | justinu | they support their rdbms on linux, and the linux os |
| 22:53.23 | mog_work | willfully ignorant |
| 22:53.25 | oogle | marcus2: fedora |
| 22:53.30 | marcus2 | heh |
| 22:53.37 | marcus2 | the windows of the linux world |
| 22:53.49 | oogle | it has its vices |
| 22:53.57 | oogle | hence me using gentoo at home |
| 22:54.14 | marcus2 | i hardly think fedora is more 'production-ready' than gentoo |
| 22:55.32 | mog_work | depends marcus2 |
| 22:55.45 | oogle | marcus2: well one of the reasons i like Fedora/RedHat for production is for running updates, and the fact that all the releases of software in the distro are tested to work together. Gentoo, being a meta-distro has less of a guarantee that every piece works together |
| 22:55.50 | mog_work | but i would tend to agree id rather drop in 50 fedora boxes than gentoo for a customer |
| 22:56.08 | marcus2 | oh, turning something over to a clueless customer is something else entirely |
| 22:56.08 | mog_work | but i would rather run gentoo as my production box than fedora |
| 22:56.14 | marcus2 | but not directly related to "production ready" |
| 22:56.35 | marcus2 | managing 50 rh boxes makes my head hurt |
| 22:56.56 | oogle | updates are the biggest problem because in gentoo you generally have to manually update your configuration files |
| 22:57.19 | oogle | Fedora is great for 'set-and-forget' |
| 22:58.17 | marcus2 | well, its ggreat for 'forget' |
| 22:58.19 | marcus2 | its the setting that sucks |
| 22:59.34 | *** join/#asterisk bweschke (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 23:01.35 | cnet2 | what is the recommended analog adapter? |
| 23:01.48 | *** join/#asterisk hilkiah (n=hilkiah@firewall.marpin.dm) |
| 23:02.40 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
| 23:02.50 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 23:02.51 | IronHelix | cnet- digium tdm board or anything from Sipura |
| 23:02.58 | cnet2 | ok |
| 23:02.59 | cnet2 | thanks |
| 23:03.09 | patpatnz | question, can I get a list of the current variables on the stack of a channel? |
| 23:03.27 | wunderkin | show channel |
| 23:03.30 | hilkiah | hi all |
| 23:03.40 | hilkiah | i have a little problem i need advice on |
| 23:03.51 | hilkiah | i have a tdm422 card which is working 75% |
| 23:04.00 | hilkiah | 1 of the fxo modules isn't working |
| 23:04.11 | mog_work | whats wrong? |
| 23:04.16 | *** part/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 23:04.20 | hilkiah | however, neither asterisk nor zttool report any problem |
| 23:04.22 | hilkiah | basically.... |
| 23:04.27 | hilkiah | the line is dead |
| 23:04.35 | hilkiah | i have 2 phone providers, A and B |
| 23:04.41 | hilkiah | and one line of each |
| 23:04.47 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:04.51 | hilkiah | so fxo ch1 = provider A and fxoch2 = provider B |
| 23:05.04 | hilkiah | say fxo ch1 does not work |
| 23:05.17 | hilkiah | if i call the number associated with this, nothing (dead air) |
| 23:05.35 | hilkiah | if I swap lines A and B the module still does not respond |
| 23:05.45 | patpatnz | wunderkin, thanks :) |
| 23:05.53 | hilkiah | i hope u guys get the drift |
| 23:06.08 | patpatnz | my client sends a Remote-Peer-ID header, does asterisk use this? |
| 23:06.13 | hilkiah | are there any tools which i can use to determine whether the fxo modules are 100% functional? |
| 23:06.37 | hilkiah | any help would be greatly appreciated |
| 23:06.40 | justinu | remote-peer-id? or remote-party-id? |
| 23:06.48 | patpatnz | peer |
| 23:06.49 | *** join/#asterisk mwright1night (n=mwright1@203-214-57-58.dyn.iinet.net.au) |
| 23:06.57 | patpatnz | is it party? |
| 23:07.12 | patpatnz | er |
| 23:07.14 | IOscanner | Okay I have rebuild a 2.6.12.3 kernel with kernel source to match and I still get this: WARNING: Symbol version dump /usr/src/linux-2.6.12.3/Module.symvers is missing; modules will have no dependencies and modversions. |
| 23:07.16 | justinu | i've never heard of remote-peer-id |
| 23:07.17 | patpatnz | sorry, remote-party-id |
| 23:07.21 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 23:07.23 | justinu | yes, asterisk can use that |
| 23:07.29 | IOscanner | I can't get the zaptel drivers to compile |
| 23:07.36 | patpatnz | justinu, but there is no var in the stack? |
| 23:07.46 | IOscanner | also: linux/autoconf.h: No such file or directory |
| 23:08.01 | patpatnz | I'm using 1.2.0-beta2 |
| 23:08.10 | rculp | I'm still having a weird issue where asterisk drops calls if they're not answered after 20 seconds |
| 23:08.18 | rculp | anyone know where that setting is hiding? |
| 23:09.21 | justinu | patpatnz: set "trustrpid=yes" in your sip.conf entry |
| 23:09.29 | wunderkin | rculp, are you using absolutetimeout? i would think that should only take over after answer |
| 23:09.37 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 23:10.07 | patpatnz | justinu, I have that set already |
| 23:10.16 | lesouvage | hilkiah: xorcom-rapid has some pretty usefull tools in the menu to test/monitor the hardware. You could download the iso, put an old hd in the box and try it out. start the menu with rapid-menu on the linux prompt. |
| 23:10.22 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:10.31 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-57.cybersurf.com) |
| 23:11.21 | hilkiah | i'll google for it |
| 23:11.39 | lesouvage | www.xorcom.com |
| 23:11.42 | patpatnz | justinu, I want to use the rpid for callerid identification |
| 23:12.20 | hilkiah | any other tools come to mind? |
| 23:12.39 | justinu | patpatnz: yeah... should work... i use it |
| 23:13.05 | justinu | patpatnz: if you want to send rremote-party-id set "sendrpid=yes" |
| 23:14.30 | patpatnz | justinu, the callerid name and number should be set to the rpid? |
| 23:14.50 | patpatnz | thats what I want it to do |
| 23:15.39 | justinu | yeah |
| 23:15.42 | IOscanner | I thought you just needed the kernel source to build asterisk. I had to drop my config for the kernel into the source tree and build the kernel half way to get asterisk to build from the kernel source. |
| 23:15.44 | justinu | sendrpid=yes |
| 23:16.20 | patpatnz | okay, will try, thanks :) |
| 23:16.42 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 23:16.49 | rculp | wunderkin: not sure |
| 23:16.54 | rculp | sorry for the slow reponse |
| 23:16.55 | patpatnz | justinu, the problem is that it doesn't use uit when making calls to h323 channels |
| 23:17.47 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
| 23:19.59 | patpatnz | justinu, also, I don't see anything about RPID in the channel variables |
| 23:22.10 | *** part/#asterisk IOscanner (n=IOscanne@216-165-210-74.crescentb.com) |
| 23:23.04 | *** join/#asterisk apardo (n=apardo@93.Red-83-43-212.dynamicIP.rima-tde.net) |
| 23:24.38 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 23:26.31 | rculp | absolutetimeout does appear to be max call length related |
| 23:26.53 | rculp | I just see stuff like |
| 23:26.59 | rculp | Nobody picked up in 20000 ms |
| 23:27.05 | rculp | Hungup 'Zap/1-1' |
| 23:27.08 | [TK]D-Fender | thats the "absolute" part kicking in ;) |
| 23:27.17 | rculp | heh |
| 23:27.23 | rculp | but if I were to get someone |
| 23:27.26 | rculp | I can talk forever |
| 23:27.28 | [TK]D-Fender | thats something you only use in pre-paid applications |
| 23:27.42 | rculp | but 20 seconds isn't enough to leave voicemail on some phone systems |
| 23:27.49 | [TK]D-Fender | no, I believe it'll kill a dia cmd... |
| 23:27.52 | wunderkin | rculp, ok well what else do you expect it to do? you need to show your dialplan, obviously theres a problem there.. |
| 23:28.04 | [TK]D-Fender | and thats why you don't need Absolutetimeout! |
| 23:28.08 | wunderkin | check the numbering |
| 23:28.10 | rculp | I had an awesome helper with my dialplan |
| 23:28.14 | rculp | ;) |
| 23:28.21 | wunderkin | people make mistakes |
| 23:28.25 | rculp | I'll post it |
| 23:28.26 | [TK]D-Fender | rc, set your root back to our last temp if you want me to take a look |
| 23:28.27 | rculp | one sec |
| 23:28.31 | [TK]D-Fender | but I've only got 5 mins |
| 23:28.42 | rculp | k |
| 23:29.03 | [TK]D-Fender | let me know when its ready |
| 23:29.10 | rculp | ready |
| 23:29.35 | rculp | the strange thing is that I don't even see where the 20 second timeout is set :) |
| 23:29.45 | [TK]D-Fender | not ready, pass isn't right |
| 23:30.04 | JonR800 | rculp: are you specifying a timeout in dial?? sorry im just jumping into this without reading much. |
| 23:30.06 | rculp | reset your pass |
| 23:30.20 | rculp | jonR: nope |
| 23:30.27 | [TK]D-Fender | ok, better |
| 23:30.48 | [TK]D-Fender | ok, what context? |
| 23:30.56 | rculp | full |
| 23:31.07 | [TK]D-Fender | which one is dying? |
| 23:31.10 | [TK]D-Fender | outgoing call? |
| 23:31.14 | rculp | yes |
| 23:31.24 | rculp | oh |
| 23:31.27 | rculp | I see the value |
| 23:31.29 | rculp | lol |
| 23:31.37 | rculp | Dial(Zap/g1/${EXTEN:1},20) |
| 23:32.05 | rculp | I r teh smart |
| 23:32.06 | JonR800 | :) i win! |
| 23:32.15 | *** join/#asterisk bbsf (n=bill@adsl-68-127-158-238.dsl.pltn13.pacbell.net) |
| 23:32.16 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 23:32.33 | X-Files | ppls, please help |
| 23:32.47 | X-Files | I have " Internet Telephony Gateway " from it I can to call on asterisk and to type numbers from ports FXS and FXO, but the problem that asterisk cannot transfer dtmf in port FXS and FXO when there is a long hooter |
| 23:33.39 | [TK]D-Fender | try now |
| 23:33.59 | rculp | k |
| 23:34.02 | [TK]D-Fender | and I got rid of that silly Zap reference in the Dial cmd! ;) |
| 23:34.23 | [TK]D-Fender | 40000ms?! |
| 23:34.25 | [TK]D-Fender | wtf? |
| 23:34.39 | rculp | it was 20000 |
| 23:34.41 | rculp | or 20 |
| 23:34.54 | rculp | I commented out that section though |
| 23:35.32 | [TK]D-Fender | ok, 1 more shot |
| 23:35.38 | [TK]D-Fender | then I've got to go for a bit. |
| 23:36.03 | rculp | I think that took care of it |
| 23:36.07 | rculp | you removing the ,20 |
| 23:36.12 | rculp | and then the ,40 that I changed it to |
| 23:36.18 | [TK]D-Fender | :O |
| 23:36.21 | [TK]D-Fender | all beter? |
| 23:36.24 | rculp | aye |
| 23:36.28 | [TK]D-Fender | ywc :) |
| 23:36.29 | rculp | now at 45 seconds |
| 23:36.32 | rculp | and counting |
| 23:36.49 | lidl | how does asterisk interface with teles gateways? |
| 23:37.01 | rculp | ty |
| 23:37.02 | [TK]D-Fender | Should stay indefinately (depending on the other side + telco) |
| 23:37.10 | [TK]D-Fender | ok, gtg for now... |
| 23:37.11 | [TK]D-Fender | bbiab |
| 23:37.12 | rculp | yeah, 1:28 now |
| 23:37.13 | [TK]D-Fender | logging out |
| 23:37.18 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 23:37.20 | [TK]D-Fender | I think you can hang up now :) |
| 23:37.36 | [TK]D-Fender | later |
| 23:41.23 | *** join/#asterisk hhoffman (n=hhoffman@71-37-17-223.tukw.qwest.net) |
| 23:41.27 | X-Files | You do not wish to help me, I have " Internet Telephony Gateway " from it I can to call on asterisk and to type on local numbers from ports FXS and FXO, but the problem that asterisk cannot transfer dtmf in ports FXS and FXO when I wish to type figures on phone all of them ignore. In what a problem? |
| 23:41.29 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 23:45.02 | *** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk) |
| 23:49.17 | *** join/#asterisk greyhound4334 (n=john@adsl-69-106-241-168.dsl.pltn13.pacbell.net) |
| 23:50.10 | X-Files | drumkilla: Russell u there ? |
| 23:50.37 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 23:51.26 | denon | X-Files: pretty sure he already left for vegas.. wont be back till next week |
| 23:51.38 | *** join/#asterisk aaronz (n=aaronz@pdpc/supporter/student/aaronz) |
| 23:52.04 | X-Files | :( |
| 23:52.21 | denon | X-Files: if you have Qs, im sure someone on the mailing list could help you |
| 23:52.45 | docelm0 | YIPPIE! |
| 23:53.54 | X-Files | It is not assured, if already nobody wrote and do not know! |
| 23:53.57 | X-Files | denon |
| 23:54.13 | greyhound4334 | hi gang. newbie, so go easy please ;-) Anyone help with strange problem: zap channel ANSWERING on calls from analog phone |
| 23:54.27 | file | plainvoip... I know who runs that... |
| 23:55.10 | greyhound4334 | That is, answering on OUTBOUND calls from analog phone sharing pstn line with x100p card |
| 23:55.28 | aaronz | does an ast_frame have to have a particular length? im trying to write an app that bi-directionally streams the audio to a java server. I should be just switching quickly between reading & writing small 1-2ms wide frames, right? |
| 23:55.46 | file | aaronz: audio usually comes in at 20ms... |
| 23:55.49 | aaronz | im assuming an ast_frame is indepentent of its format, right? |
| 23:56.33 | aaronz | file: how would i create a low-latency full-duplex channel to a local server? |
| 23:56.47 | X-Files | denon: You concern to asterisk programming? |
| 23:56.56 | aaronz | has to be simultaneous, i cant drop every other 20ms |
| 23:56.59 | denon | no, I'm just in marketing |
| 23:57.09 | file | aaronz: there is an icecast app... |
| 23:57.30 | aaronz | how low-latency is that? |
| 23:57.43 | X-Files | denon: :) hmm, i how see in cvs update by Russell 17 min back |
| 23:57.55 | file | yeekz my other LCD needs to be cleaned |
| 23:57.55 | file | ACHOO |
| 23:58.01 | PlainVoip-DocE | file huh? |
| 23:58.11 | X-Files | denon: You are assured hundred it are not present? |
| 23:58.24 | aaronz | file: and thats one way then |
| 23:58.26 | denon | que? |
| 23:58.26 | aaronz | not full duplex |
| 23:58.45 | file | plainvoip is a company |
| 23:59.31 | docelm0 | Yes |
| 23:59.38 | denon | X-Files: I dont know for sure .. but I dont think he is around now |
| 23:59.39 | docelm0 | Well subsidiary of another.. but yes |
| 23:59.53 | denon | X-Files: besides, drumkilla's english isnt so good .. he doesnt like to talk on irc much |
