| 00:00.09 | lters_ | DaeJeon-novice: if you only have one partion there, than in menu.lst add |
| 00:00.13 | DaeJeon-novice | result http://pastebin.ca/281704 |
| 00:00.36 | DaeJeon-novice | I have two disks |
| 00:00.43 | DaeJeon-novice | hda -centos |
| 00:00.47 | DaeJeon-novice | hdb debian |
| 00:01.51 | matt_ | you have 2 disks and you cant "cd /boot/grub/" |
| 00:01.59 | matt_ | 2 disks of linux os's |
| 00:02.05 | DaeJeon-novice | yes |
| 00:02.07 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 00:02.12 | matt_ | ok |
| 00:02.49 | lters_ | do this, "mkdir /mnt/hda1" then "mount /dev/hda1 /mnt/hda1" than "ls /mnt/hda1/" |
| 00:03.01 | lters_ | that should show the cent os kernel name |
| 00:03.36 | lters_ | matt_ are you on ipv6? |
| 00:03.45 | matt_ | lters, yea |
| 00:03.53 | lters_ | thats fancy |
| 00:04.56 | DaeJeon-novice | lters_: yes |
| 00:04.59 | DaeJeon-novice | it did |
| 00:05.03 | matt_ | lters_, i have a firefox plugin that shows the remote ip of the web server, and if its ipv4 its red and ipv6 its green, theres quite a few sites these days that have ipv6 support |
| 00:05.08 | DaeJeon-novice | I can see the kernelname now |
| 00:05.54 | lters_ | http://pastebin.ca/281711 |
| 00:06.18 | lters_ | DaeJeon-novice: add that to the bottom of the menu.lst , save and reboot |
| 00:07.06 | matt_ | cant you just leave out the initrd line ? |
| 00:07.18 | lters_ | yeah, sorry, leave that blank |
| 00:07.34 | DaeJeon-novice | result of ls /mnt/hda1/ ---http://pastebin.ca/281712 |
| 00:07.58 | matt_ | unless your kernel dosn't have support for your root filesystem ... |
| 00:07.58 | matt_ | lol |
| 00:08.35 | lters_ | DaeJeon-novice: so, where it says put kernel here, put this instead vmlinuz-2.6.9-42.0.3.EL |
| 00:09.07 | DaeJeon-novice | other things same? |
| 00:09.08 | matt_ | something isn't right |
| 00:09.25 | matt_ | in grub.conf you have .... /boot/kernelnamehere |
| 00:09.38 | matt_ | but a ls /mnt/hda1 shows the kernel is in / on that filesystem |
| 00:09.42 | matt_ | and not /boot |
| 00:10.18 | lters_ | I was thinking about that too. remove the /boot from that 3rd line. |
| 00:10.22 | matt_ | if you did ls /mnt/hda1/boot and got the kernel that would be fine |
| 00:10.45 | matt_ | hda1 is your boot partition, i dont see any /usr /proc /dev |
| 00:10.56 | matt_ | im guessing hda2 is your root partition |
| 00:11.07 | matt_ | so ... kernel /boot/kernelnamehere root=/dev/hda1 ro .. should be .. |
| 00:11.15 | matt_ | kernel /kernelnamehere root=/dev/hda2 ro |
| 00:11.30 | *** join/#asterisk JJMan123 (n=justin@5ac032e4.bb.sky.com) |
| 00:12.19 | DaeJeon-novice | i am confused |
| 00:12.24 | matt_ | but then hda2 might be your swap |
| 00:12.37 | matt_ | i and hda3 your root |
| 00:12.39 | DaeJeon-novice | can give the file code to put in |
| 00:12.43 | matt_ | wheres your fdisk ? |
| 00:12.46 | matt_ | output |
| 00:13.10 | lters_ | http://pastebin.ca/281711 |
| 00:13.19 | matt_ | LOL |
| 00:13.20 | matt_ | /dev/hda2 14 2438 19478812+ 8e Linux LVM |
| 00:13.23 | matt_ | linux LVM |
| 00:13.32 | matt_ | your root is on a logical volume manager |
| 00:14.03 | matt_ | that stuff is WAY more complicated to boot off than just a normal partition with a fs |
| 00:14.42 | DaeJeon-novice | how can I do? |
| 00:14.43 | matt_ | it might be a better idea to leave out LVM support and play around with /boot grub and getting to know filesystems n stuff |
| 00:15.06 | matt_ | that might be what the initrd was setting up |
| 00:15.33 | matt_ | when you installed centos ... was the drive connected to your pc in the same way it is now? |
| 00:15.44 | DaeJeon-novice | yes |
| 00:15.51 | matt_ | ok |
| 00:16.18 | DaeJeon-novice | centos drive is primary master |
| 00:16.24 | matt_ | yea |
| 00:16.31 | DaeJeon-novice | debian primary s |
| 00:16.39 | lters_ | there is a img out there that should work. |
| 00:16.46 | matt_ | when you boot you get a debian boot? |
| 00:16.58 | DaeJeon-novice | yes |
| 00:17.08 | DaeJeon-novice | only debian |
| 00:17.19 | matt_ | ok your booting from hdc the drive with debian on |
| 00:17.32 | matt_ | and all the modifaications that you are making are on hda |
| 00:17.48 | matt_ | so even if this works you will still need to tell your bios to boot from the different drives |
| 00:17.54 | matt_ | to boot into the different os's |
| 00:18.42 | matt_ | if you want 2 menu options then you will need to edit the menu.lst on the debian /boot partition |
| 00:19.05 | DaeJeon-novice | what should I put in? |
| 00:19.16 | lters_ | http://pastebin.ca/281720 |
| 00:19.55 | DaeJeon-novice | final? |
| 00:20.05 | DaeJeon-novice | good to go? |
| 00:20.18 | lters_ | I would try that, unless matt_ sees any catches.. |
| 00:20.43 | lters_ | cya |
| 00:20.56 | matt_ | the only thing i can see is because your using lvm |
| 00:21.04 | matt_ | your root might not be /dev/hda2 |
| 00:21.18 | DaeJeon-novice | can't we check that? |
| 00:21.26 | matt_ | but something like /dev/group01/lvm01 |
| 00:21.30 | matt_ | just remove it |
| 00:21.51 | matt_ | http://pastebin.ca/281725 |
| 00:21.58 | matt_ | the initrd should take care of that for you |
| 00:22.40 | DaeJeon-novice | final? |
| 00:23.09 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 00:23.11 | matt_ | yea, you will have to tell your bios to boot from hda |
| 00:23.26 | matt_ | but if it works you can put it in the debians menu.lst |
| 00:24.58 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 00:25.00 | drfreeze | Hi |
| 00:25.02 | DaeJeon-novice | yes system will boot from hda |
| 00:25.20 | DaeJeon-novice | I already changed the bios |
| 00:25.49 | drfreeze | I've got a single polycom phone that has decided it wants to show the time 6 hrs behind. (the gmt offset is ok) |
| 00:25.57 | drfreeze | Any ideas why this would happen? |
| 00:27.33 | *** join/#asterisk brian (i=brian@unaffiliated/brian) |
| 00:38.43 | DaeJeon-novice | matt_: did not work |
| 00:39.00 | DaeJeon-novice | could not mount |
| 00:46.21 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:47.19 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 00:48.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:57.20 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:57.20 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.14, 1.4.0-beta4, Zaptel 1.2.12, 1.4.0-beta3 Released! (Dec. 15, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 01:00.45 | *** join/#asterisk ast_freak|Laptop (n=jesse@dhcp.208148.en-tel.net) |
| 01:15.52 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 01:15.52 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.14, 1.4.0-beta4, Zaptel 1.2.12, 1.4.0-beta3 Released! (Dec. 15, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 01:22.23 | *** join/#asterisk Omer (i=Omer@203.81.233.202) |
| 01:31.16 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:47.25 | *** join/#asterisk hbsmurf (n=ghandi@68-188-139-162.dhcp.aldl.mi.charter.com) |
| 01:47.28 | hbsmurf | Howdy |
| 01:48.17 | hbsmurf | Anyone here know if multi-server asterisk installs are a pain in the ass when it comes to cdr? |
| 01:52.10 | Nugget | as long as you're using a database-backed cdr (like cdr_odbc or cdr_pgsql) I'd expect you'll have no problems. |
| 01:52.49 | hbsmurf | ok |
| 01:52.49 | hbsmurf | figured as much |
| 01:52.49 | hbsmurf | thanks |
| 01:52.49 | hbsmurf | Wasn't sure anyone would be here on a Saturday night |
| 01:52.49 | hbsmurf | but figured I'd try |
| 01:52.50 | hbsmurf | :) |
| 01:52.51 | Younss | hehe |
| 01:52.53 | Younss | hi |
| 01:53.01 | hads | It's not Saturday everywhere mate. |
| 01:53.24 | hbsmurf | Yeah yeah |
| 01:53.33 | hbsmurf | it could be Sunday morning |
| 01:53.34 | hbsmurf | :) |
| 01:53.54 | hads | Or Sunday afternoon. |
| 01:54.04 | hbsmurf | Good point |
| 01:54.05 | hbsmurf | :) |
| 01:54.38 | *** join/#asterisk doolph (n=doo@200.46.148.58) |
| 01:54.51 | doolph | sup |
| 01:54.56 | hbsmurf | supsup |
| 01:55.11 | hbsmurf | I've gotta shut the power off so I can get my new outlets hooked up |
| 01:55.22 | hbsmurf | got the plywood on the wall in my mechanical room |
| 01:55.33 | hbsmurf | I need a decent shelf up there for my server |
| 01:55.41 | hbsmurf | and my wife doesn't think I'm crazy |
| 01:57.08 | lters_ | DaeJeon-novice: and ? |
| 02:13.07 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 02:16.58 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:18.21 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.159.213) |
| 02:18.39 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
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| 03:00.40 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
| 03:00.46 | Mad|Cow | anyone know if there is such as thing as a tftp resource record? I'm trying to figure out how to make my phone find my tftp server without hard coding the IP. |
| 03:02.29 | naftali5 | there is a DHCP option for that you can set up in your router sometimes |
| 03:02.35 | naftali5 | don't know much more |
| 03:03.17 | Mad|Cow | I was hoping you could add a service record or resource record.... but I cant find anything on it... not sure if such a thing exists |
| 03:03.26 | JT | Mad|Cow: you need to set it at the dhcp server |
| 03:03.40 | JT | you can set arbitary information in a decent DHCPd |
| 03:04.00 | Mad|Cow | JT: Do you know what the record needs to say? |
| 03:04.40 | JT | no, read your phone documentation |
| 03:05.18 | naftali5 | http://www.google.com/search?hl=en&q=dhcp+option+tftp&btnG=Google+Search |
| 03:06.46 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 03:06.48 | Mad|Cow | naftali5: you rock... I have been searching all over google... couldnt find a thing :-) |
| 03:10.08 | *** join/#asterisk dmd1222 (n=opera@c-24-20-35-49.hsd1.or.comcast.net) |
| 03:15.18 | *** join/#asterisk nvicf (n=v@201.250.183.191) |
| 03:17.32 | *** part/#asterisk dmd1222 (n=opera@c-24-20-35-49.hsd1.or.comcast.net) |
| 03:24.37 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 03:24.41 | Stephnie | hi |
| 03:25.33 | Stephnie | when I issue "sip reload" ... then it takes a bit time to reload "sip_notify.conf" ....any help?? |
| 03:26.55 | naftali5 | dns issues |
| 03:27.03 | Stephnie | u r right... |
| 03:27.25 | Stephnie | I got some msgs last nite .... unable to look dns |
| 03:27.29 | naftali5 | check your /var/log/asterisk/full you should see the 20 second pauses on DND lookup |
| 03:27.51 | naftali5 | try to ping each of your provider hostnames on py one |
| 03:27.57 | Stephnie | is this issue with my dns or the dns my asterisk is registring with ? |
| 03:28.00 | naftali5 | *DNS lokups |
| 03:28.09 | naftali5 | *lookups |
| 03:28.25 | Stephnie | :) |
| 03:28.49 | Stephnie | so this problem is with any of my service provider's dns ? |
| 03:29.19 | naftali5 | maybe them, maybe your internet, maybe your ISP's DNS server |
| 03:29.29 | naftali5 | like i said, try to ping them one by one |
| 03:29.59 | Stephnie | okey...let me check plz |
| 03:30.17 | naftali5 | check the logs, you should see the one/ones which made it pause around 20 seconds |
| 03:30.47 | Stephnie | cdr-csv cdr-custom event_log messages queue_log |
| 03:31.15 | Stephnie | that is what I have in /var/log/asterisk |
| 03:34.21 | nvicf | I've configured extension, sip, and the rest of asterisk, I'm restarting it, and when I try to use a softphone from another machine it doesn't connect, I've nmaped the machine and it doesn't have anything open, is that correct? |
| 03:41.59 | *** part/#asterisk [Airwolf] (n=airwolf@89.205.159.213) |
| 03:42.58 | *** part/#asterisk Bog (n=Bog@CPE00179a9ca2b5-CM00080d7be684.cpe.net.cable.rogers.com) |
| 03:51.23 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:52.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:52.43 | *** join/#asterisk bmg505 (n=leon@c1-233-8.rndf.isadsl.co.za) |
| 03:54.12 | shmaltz | anybody here seen the news? I'm Time Magazine Person of the year |
| 03:56.52 | Stephnie | naftali5: if I get ping reply from all of the domains/dns of my service providers then ? |
| 03:57.31 | Stephnie | naftali5: but I am sure ...problem is with dns...as asterisk used to prompt me unable to lookup sip.broadvoice.com |
| 03:58.15 | naftali5 | maybe at specific times |
| 03:58.31 | naftali5 | but if it prompts you only for one, and you have multiple |
| 03:58.38 | naftali5 | it is probably their problem |
| 03:58.59 | Stephnie | but they are working fine at my another dedicated server.... |
| 03:59.40 | Stephnie | must be some problem with my current dedicated server....and their tech says....there is not any problem at their end. |
| 03:59.58 | *** join/#asterisk suma (n=suma@cm136.omega182.maxonline.com.sg) |
| 04:00.17 | suma | any place where i can get iax DID's for US? if possible free |
| 04:01.42 | Stephnie | naftali5: they asked me to check dig command |
| 04:02.16 | shmaltz | looks interesting: |
| 04:02.17 | shmaltz | http://news.yahoo.com/s/cmp/20061214/tc_cmp/196604025 |
| 04:04.08 | naftali5 | Stephanie: I was able to pinpoint your problem because I have seen it before, that's where my expertise ends |
| 04:04.36 | Stephnie | :) thanks |
| 04:05.27 | *** join/#asterisk obnauticus (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 04:15.40 | nvicf | I'm having the obnoxius Found no files in '/usr/share/asterisk/mohmp3', where can I find those files? |
| 04:16.30 | dlynes_laptop | ftp://ftp.digium.com/pub/telephony/sounds |
| 04:17.10 | nvicf | thanks godines |
| 04:17.20 | dlynes_laptop | who's godines? |
| 04:18.01 | nvicf | a funny character from a mexican comedy, el chavo del 8 |
| 04:18.08 | nvicf | dlynes sounded like godines in my head |
| 04:18.13 | dlynes_laptop | ic |
| 04:18.14 | Stephnie | Dec 16 19:08:15 WARNING[16251]: chan_sip.c:1989 create_addr: No such host: sip.broadvoice.com |
| 04:18.14 | Stephnie | Dec 16 19:08:15 WARNING[16251]: chan_sip.c:5470 transmit_register: Probably a DNS error for registration to xxxxxxxxxx6@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) |
| 04:18.25 | nvicf | dlynes_laptop, how do I know which one of those? |
| 04:18.26 | nvicf | any? |
| 04:18.43 | Stephnie | damn! |
| 04:18.46 | dlynes_laptop | nvicf: Just grab all the .wav files |
| 04:19.17 | Stephnie | I am stucked in 2 service providers...both says...they are fine...then I must be wrong in paying that b**** |
| 04:19.30 | dlynes_laptop | Stephnie: Just put in 147.135.12.128 instead of sip.broadvoice.com if you're having issues |
| 04:20.43 | dlynes_laptop | Stephnie: that'll solve the issue until your dns server gets fixed |
| 04:21.05 | dlynes_laptop | Stephnie: then when you're able to type in 'dig sip.broadvoice.com' and it works |
| 04:21.15 | dlynes_laptop | Stephnie: change it back to sip.broadvoice.com |
| 04:21.23 | *** join/#asterisk De_Mon (n=de_mon@fl-76-4-98-162.dhcp.embarqhsd.net) |
| 04:21.42 | Stephnie | dlynes_laptop: I think I tried that ..but let me check again |
| 04:22.06 | dlynes_laptop | Stephnie: obviously you didn't, or it'd be sayiing no such host: 147.135.12.128 instead |
| 04:23.18 | dlynes_laptop | Stephnie: after you made your changes, did you do a 'sip reload'? |
| 04:24.41 | Stephnie | yes... |
| 04:24.45 | Stephnie | checking again |
| 04:25.05 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 04:29.02 | *** join/#asterisk frogzoo_ (n=frogzoo@202.155.165.25) |
| 04:35.34 | *** join/#asterisk ipl31 (n=ken@h-67-101-91-165.sttnwaho.covad.net) |
| 04:42.52 | *** join/#asterisk saftsack (n=saftsack@pD9E06112.dip.t-dialin.net) |
| 04:43.22 | linagee | has anyone played with that wengo flash phone? |
| 04:43.27 | linagee | does it work with asterisk? |
| 04:43.42 | linagee | it says SIP compatible.... |
| 04:44.42 | nvicf | weird, I'm downloading the spanish files and still sounds in english! |
| 04:44.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 04:45.07 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 04:45.20 | FuriousGeorge | nvicf: how can spanish sound english? |
| 04:45.44 | nvicf | FuriousGeorge, no idea, I've downloaded the english files, then deleted everything, and downloaded the spanish files |
| 04:47.51 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 04:49.48 | resistance | could anyone recommend a sound board to use for the intercom system |
| 04:50.11 | resistance | i have a sound blaster and theres a horrible sound which sounds like an echo |
| 04:56.10 | linagee | FuriousGeorge: maybe it has an english accent? :) |
| 04:57.23 | nvicf | no man, that's english |
| 04:57.53 | FuriousGeorge | nvicf: are you using a@h? |
| 04:58.00 | nvicf | no, only asterisk |
| 04:58.28 | FuriousGeorge | nvicf: and you changed the filenames in the dialplan? |
| 04:58.37 | nvicf | I've downloaded ../asterisk-core-sounds-es-wav-1.4.4.tar.gz ../asterisk-moh-freeplay-wav.tar.gz |
| 04:58.37 | nvicf | ../asterisk-core-sounds-es-wav-current.tar.gz |
| 04:58.40 | dlynes_laptop | resistance: maybe lower your input gain on the soundblaster? |
| 04:58.50 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 04:59.12 | dlynes_laptop | nvicf: you need either 1.4.4 or the current, but not both |
| 04:59.38 | nvicf | one overwrites the other I guess |
| 05:00.06 | FuriousGeorge | so the spanish filenames are the same as the english names? |
| 05:00.30 | nvicf | appears to be when I detargz |
| 05:01.18 | FuriousGeorge | in a way that makes sense, but what if you wanted to have both on your system, which to me makes just as much sense, you would obviously need different files |
| 05:01.57 | nvicf | mmm |
| 05:03.01 | FuriousGeorge | nvicf: having said that, it should be very simple. try to find on your filesystem the spanish clip you are expecting to get |
| 05:03.16 | FuriousGeorge | either its there, and you can invoke it from the dialplan, or it's not |
| 05:03.20 | FuriousGeorge | and you cant |
| 05:04.23 | resistance | dlynes-laptop: how can i do that? |
| 05:04.42 | FuriousGeorge | resistance: is this a usb mic? |
| 05:04.58 | FuriousGeorge | err, you already said it was your soundcard |
| 05:04.59 | FuriousGeorge | nm |
| 05:05.07 | resistance | no, i dial into the console with my phone |
| 05:05.12 | FuriousGeorge | analog phone? |
| 05:05.16 | resistance | yes |
| 05:05.22 | FuriousGeorge | txgain/rxgain |
| 05:05.29 | resistance | 0.0 |
| 05:05.31 | resistance | bith |
| 05:05.35 | resistance | both |
| 05:05.40 | FuriousGeorge | so try negative values |
| 05:06.08 | FuriousGeorge | tx is transmit and rx is receive in case you didnt know |
| 05:06.18 | FuriousGeorge | i forget what the range is |
| 05:06.19 | resistance | yeah |
| 05:06.31 | FuriousGeorge | +/-10? |
| 05:07.24 | resistance | it seems that my voice when i speak into the phone breaks up, it really funny |
| 05:07.37 | *** join/#asterisk bulatitoy (n=r@adsl-70-231-146-111.dsl.snfc21.sbcglobal.net) |
| 05:07.40 | resistance | it doesn't matter how much i have the volume turned up or down |
| 05:08.00 | resistance | but i will try the gain |
| 05:08.00 | FuriousGeorge | did you reload chan_zap after changing the values |
| 05:08.11 | resistance | thanks 4 the advice |
| 05:08.12 | FuriousGeorge | you might even have to restart entrirely |
| 05:08.13 | bulatitoy | need help regarding zaptel compile |
| 05:08.20 | resistance | no, i'm actually not at my box right now |
| 05:08.44 | nvicf | I have no idea where this come from |
| 05:09.00 | nvicf | I've tar-zxvf in /usr/share/asterisk/mohmp3 |
| 05:09.05 | nvicf | mmm maybe it doesn't go there |
| 05:09.20 | FuriousGeorge | nvicf: show me the ls output of the file you expect it to play |
| 05:09.50 | bulatitoy | its says in thebook before compiling zaptel you need to do ln -s to point to the kernel source |
| 05:09.52 | nvicf | -rw-r--r-- 1 arquimedes arquimedes 631698 2006-12-11 19:48 demo-congrats.wav |
| 05:10.09 | bulatitoy | how do you do that in debian sarge? |
| 05:10.33 | FuriousGeorge | nvicf: doesnt it go in /var/lib/asterisk/sounds/ |
| 05:11.24 | FuriousGeorge | bulatitoy: when you type ls -la in /usr/src do you see a link called linux pointing to a dir called linux-(something)-2.6.(something)? |
| 05:11.52 | nvicf | FuriousGeorge, what?but it complains about mohmp3 if it's empty |
| 05:13.31 | FuriousGeorge | nvicf: if you were your client wouldnt you complain if you had no Music On Hold b/c someone hadnt put any MP3s in that dir |
| 05:13.34 | nvicf | I though it was a different thing |
| 05:13.50 | nvicf | FuriousGeorge, I have no /var/lib/asterisk/sounds |
| 05:13.56 | nvicf | FuriousGeorge, only /var/lib/asterisk |
| 05:14.09 | nvicf | FuriousGeorge, I have /usr/share/asterisk/sounds/ |
| 05:14.13 | FuriousGeorge | nvicf: so you dont have asterisk-sounds installed |
| 05:14.15 | nvicf | but I have some gsm I don't want to screw |
| 05:14.20 | bulatitoy | Furious: no i dont |
| 05:14.27 | FuriousGeorge | so by all means dont screw them :) |
| 05:14.48 | FuriousGeorge | bulatitoy: do you have that linux-(something)-2.6.(something) |
| 05:15.05 | nvicf | FuriousGeorge, perfect it goes in /usr/share/asterisk/sounds, btw spanish sounds are really bad:) |
| 05:15.27 | bulatitoy | do i need to do ln -s /boot/config-2.4 /usr/src/linux-2.4? |
| 05:15.28 | FuriousGeorge | nvicf: tiene us accento terrible? |
| 05:15.31 | FuriousGeorge | *un |
| 05:15.46 | nvicf | FuriousGeorge, lol, it sounds like arnold speaking english |
| 05:15.55 | FuriousGeorge | bulatitoy: close |
| 05:16.06 | bulatitoy | i have the config-2.4 in /boot |
| 05:16.08 | FuriousGeorge | bulatitoy: you need to go into /usr/src (cd /usr/src) |
| 05:16.11 | bulatitoy | is that the file? |
| 05:16.31 | FuriousGeorge | bulatitoy: err, it should be a dir with kernel headers installed |
| 05:16.39 | bulatitoy | all i have in /usr/src are the asterisk source |
| 05:16.57 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 05:17.03 | FuriousGeorge | bulatitoy: in debian, it sounds like you need to apt-get your kernel headers |
| 05:17.23 | bulatitoy | i see |
| 05:17.28 | FuriousGeorge | nvicf: its not allison, is it? |
| 05:17.38 | bulatitoy | the kernel-headers are already 2.6 |
| 05:17.46 | bulatitoy | my kernel version is only 2.4 |
| 05:17.49 | bulatitoy | is that fine? |
| 05:17.54 | FuriousGeorge | no |
| 05:18.15 | bulatitoy | so i need to upgrade my kernel to 2.6 |
| 05:19.06 | FuriousGeorge | bulatitoy: if you have linux kernel headers installed why dont you have a linux-2.6 something? |
| 05:20.02 | FuriousGeorge | bulatitoy: truth is, i use gentoo which "comes with" kernel headers by default, so im not exactly sure what package you need |
| 05:20.24 | bulatitoy | i see...i will try gentoo later |
| 05:20.24 | FuriousGeorge | but you could just upgrade to 2.6 wich i think everyone will agree plays nicer with * |
| 05:20.37 | FuriousGeorge | bulatitoy: im not suggesting you use gentoo |
| 05:20.51 | FuriousGeorge | :) |
| 05:21.11 | bulatitoy | i have tried trixbox earlier but with dialing out is inconsistent, am using x100p to test |
| 05:21.20 | FuriousGeorge | if you think you can do it, upgrade to 2.6 |
| 05:21.51 | bulatitoy | yeah i think thats the best way to do it |
| 05:21.53 | nvicf | FuriousGeorge, no idea, it doesn't show any names |
| 05:21.58 | FuriousGeorge | never used an x100p but i hear people have bad luck with it |
| 05:22.08 | FuriousGeorge | nvicf: she is the voice of the default sounds |
| 05:22.10 | bulatitoy | i really wanted to make it work on debian |
| 05:22.45 | bulatitoy | yeah, ive read a lot about x100p, shouldnt hve bought it, tho its on $10 |
| 05:22.56 | nvicf | FuriousGeorge, ah no idea, I can only use 8bit mp3 for music on hold isn't? |
| 05:23.20 | FuriousGeorge | bulatitoy: and if you want "production quality" analog channels use a server mb. zaptel hw is sensitive about irqs |
| 05:23.38 | FuriousGeorge | and maybe upgrade to tdm400 :) |
| 05:23.49 | bulatitoy | are the TDM400P cards good? |
| 05:23.57 | FuriousGeorge | nvicf: i believe so |
| 05:24.04 | bulatitoy | how about the sangoma? |
| 05:25.28 | danp | i'm liking my sangoma so far |
| 05:25.51 | bulatitoy | is it more expensive than the tdm400p? |
| 05:27.42 | danp | how many portts do you need? |
| 05:30.16 | danp | a quick froogle search and digging got me a tdm400p with 4 FXO ports for $395. the comparable sangoma (A20002) is $360 |
| 05:30.42 | danp | but i'm sure those prices vary quite a bit from place to place |
| 05:31.19 | bulatitoy | i see |
| 05:31.20 | [TK]D-Fender | And the A200 can expan past 4 ports without requiring more PCI resources and at a lower cost. |
| 05:31.25 | danp | true |
| 05:31.37 | danp | i have a 12 FXO port setup |
| 05:31.39 | bulatitoy | how about echo cancellation? |
| 05:31.48 | bulatitoy | is it better on sangoma? |
| 05:32.44 | [TK]D-Fender | bulatitoy : flawless in my experience. the TDM400 doesn't have any hardware EC. |
| 05:33.36 | bulatitoy | we are planning on replacing our old key tel system |
| 05:33.53 | bulatitoy | we only have 4 pstn lines and planning go get a voip service |
| 05:34.16 | bulatitoy | i would definitely look in to the sangoma card |
| 05:34.58 | bulatitoy | but im totally new to * thats why im testing everything from trixbox to the asterisknow |
| 05:35.14 | bulatitoy | but would like to make it work on debian |
| 05:38.32 | danp | i'm having no trouble using my sangoma and asterisk on an ubuntu system |
| 05:38.46 | [TK]D-Fender | bulatitoy : There are guides for FreePBX for Debian, but if its a set&forget box, why bother. |
| 05:38.59 | nvicf | damn, not getting any music on hold, I've used lame to convert to 8bit |
| 05:39.06 | [TK]D-Fender | Then again.... * GUI's ..... shudder |
| 05:39.33 | hads | Don't discount external gateways either, they give you driver independance. |
| 05:41.56 | bulatitoy | i find the GUI more difficult, its suppose to be easier :) |
| 05:42.09 | bulatitoy | especially when i tried asterisknow beta1 |
| 05:43.11 | [TK]D-Fender | The reason for choosing * is control. And that is exactly what you give up when you take that road. |
| 05:45.13 | bulatitoy | right, thats why, as much as possible, i want to make it work on debian and do the tinkering on the CLI |
| 05:45.50 | Stephnie | Asteriswin32 only supports 2 concurrent channels ...is that true? |
| 05:46.14 | [TK]D-Fender | bulatitoy : tip : don't touch any of the debian packages, just build from source on a sane intall of Debian. |
| 05:46.53 | bulatitoy | the try, i did the apt-get |
| 05:47.03 | bulatitoy | i was not able to make it work |
| 05:47.19 | bulatitoy | now im trying again w/o using deb package |
| 05:47.50 | bulatitoy | by the way, i dont have to do the ln -s thingy when im using kernel 2.6 right? |
| 05:48.40 | [TK]D-Fender | bulatitoy : No idea what you're talking about. |
| 05:48.59 | [TK]D-Fender | bulatitoy : Just make sure you have the pre-req's installed, and that you build it the way they tell you to. |
| 05:50.58 | Stephnie | dlynes_laptop: u there? |
| 05:51.05 | bulatitoy | it says in thebook that you need to make a symbolic link pointing to the kernel source before compiling zaptel |
| 05:54.05 | [TK]D-Fender | bulatitoy : You should have your kernel's source in /usr/src just try it out. yes I suppose you might have to make a link for Linux and Kernel. |
| 05:54.59 | bulatitoy | im compiling now, hope it works...tried several things today |
| 05:55.23 | bulatitoy | i was not happy with trixbox |
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| 05:57.12 | bulatitoy | one problem with trixbox and the x100p is sometimes the x100p gets the unconfigured alarm |
| 05:57.32 | bulatitoy | then u have to run genzaptelconf twice to make it work |
| 05:58.01 | nvicf | I have a voip account to connect via softphone and they didn't gave me any info as to how to connect asterisk, that's nice... there are like 30 ways according to the docs |
| 05:58.03 | bulatitoy | make a call once, it will get through (dialing my cellphone) |
| 05:58.17 | bulatitoy | try to redial, then boom, wont work |
| 05:58.48 | dlynes_laptop | Stephnie: yep |
| 05:59.00 | dlynes_laptop | Stephnie: what's up? |
| 06:06.42 | bulatitoy | just finished doing make on asterisk i received this error make: *** [editline/libedit.a] Error 1 |
| 06:06.50 | bulatitoy | what is it? |
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| 06:14.34 | Stephnie | dlynes_laptop: if I change sip.broadvoice.com to 147.135.12.128 then line doesnt get registered. |
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| 06:14.34 | Stephnie | dlynes_laptop: broadvoice have some kind of restrictions I think....I must need to use sip.broadvoice.com ...other than IP Address...I have checked it in xlite softphone as well. |
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| 06:40.10 | Lurchtoke | hey all |
| 06:40.45 | nvicf | sup lurchtoke |
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| 06:47.08 | Lurchtoke | nothing much...just trying to get my first taste of asterisk |
| 06:47.40 | Lurchtoke | taking over a company...guess who has the responsability of making sure our asterisk server stays running :/ |
| 06:47.50 | Lurchtoke | lol |
| 06:48.05 | nvicf | :S |
| 06:48.10 | nvicf | not even funny |
| 06:48.12 | nvicf | :P |
| 06:48.20 | Lurchtoke | nope |
| 06:48.24 | Lurchtoke | crash course.... |
| 06:48.35 | Lurchtoke | Im thinking the gui would be best for me |
| 06:48.37 | nvicf | what company? |
| 06:48.48 | nvicf | not air company I hope |
| 06:48.53 | Lurchtoke | nope |
| 06:49.07 | Lurchtoke | retail company with three stores |
| 06:49.19 | Lurchtoke | asterisk is co-loed at time warner |
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| 06:49.48 | Lurchtoke | shit would go down all the timedue to the idiots stupidity |
| 06:50.09 | Lurchtoke | Im thinking just delete the shit and reinstall fresh and setup with the gui |
| 06:50.27 | Lurchtoke | Ive down pascal...and some unix...but its been a while |
| 06:50.30 | Lurchtoke | err done |
| 06:50.41 | Lurchtoke | pascal...basic...unix...and a little C++ |
| 06:51.04 | nvicf | nice |
| 06:51.17 | Lurchtoke | but alas....never messed with asterisk |
| 06:51.43 | Lurchtoke | shoot....too bad its not irc scripting |
| 06:51.44 | Lurchtoke | lol |
| 06:53.31 | Lurchtoke | basic setup is a T-1 coming into our main store with 4 sip adapters......the other 2 stores gotr sdsl 768/768 |
| 06:54.08 | Lurchtoke | so....I have like 3 weeks to learn...because we are moving main store and I gotta set it up with my head tech |
| 06:54.18 | nvicf | lol |
| 06:54.32 | Lurchtoke | our company sells computers and electronics..and does computer service work |
| 06:54.52 | Lurchtoke | so....on top of that I gotta deploy everest software also |
| 06:54.55 | Lurchtoke | blah |
| 06:55.37 | nvicf | nice |
| 06:56.00 | Lurchtoke | im thinking best solution might be to hire third party to set it up |
| 06:56.06 | Lurchtoke | (shrug) |
| 06:56.29 | Lurchtoke | less stress for me |
| 06:56.33 | Lurchtoke | :P |
| 06:56.34 | nvicf | I'm in argentina so... |
| 06:56.37 | nvicf | far far |
| 06:56.43 | nvicf | cheap cheap, but far far |
| 06:57.14 | Lurchtoke | lol |
| 06:59.56 | FuriousGeorge | Lurchtoke: where are the stores? |
| 07:00.28 | coppice | nvicf: how much do the digium cards cost in .ar? I heard some extremely high prices for them in some of your neighbours, due to import tarrifs |
| 07:01.01 | nvicf | coppice, depends, which type? |
| 07:01.38 | coppice | lets say a TE405P |
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| 07:04.16 | humbug | I am trying to create a macro that executes some additional commands after a user leaves a voicemail. The problem is that after the user hangs up the commands below Voicemail() don't get executed |
| 07:05.19 | nvicf | coppice, can't get prices because the reseller is not giving those in the web, but something like, Digium X100p Clon Port Fxo Para Asterisk Voip, USD45 |
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| 07:06.51 | coppice | avicf: since that is a clone, and we don't know how it was created, it might not be representative. In brazil someone told me the T1/E1 cards are several times the US price, due to taxes |
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| 07:10.15 | nvicf | mmm no idea really, but yes, things are expensive, not only due to taxes, in argentina are worst because of 3x1 devaluation |
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| 08:27.23 | THX2000 | has anyone tried those grandstream 8x fxo gateways? |
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| 09:17.04 | qwertz | HI, just installed asterisk 1.2.13-BRIstuffed on a debian etch rc1. If I call the voicemail app the debug output shows "Playing 'vm-youhave' (language 'en')" but I don't hear anything on the phone. Using mpg123 on the linux cli I can hear mp3 so the sound system seems to work. * sounds are in /usr/share/asterisk/sounds - so is there a way to test if * tries to open the right files or are there some other things I could check? |
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| 09:37.45 | qwertz | using MP3Player also works from inside of |
| 09:38.31 | qwertz | *, so the problem seems to be the gsm file so is there any special lib needed to play it? |
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| 10:50.22 | WQDEQWDQW | hello all |
| 10:50.44 | WQDEQWDQW | anyone can help me with a trouble |
| 10:50.46 | WQDEQWDQW | ? |
| 10:51.05 | naftali5 | ? |
| 10:51.27 | WQDEQWDQW | i have some problem receiving calls with asterisk |
| 10:51.38 | WQDEQWDQW | i installed it on my wrt54gs |
| 10:52.05 | WQDEQWDQW | I make sip.conf |
| 10:52.24 | WQDEQWDQW | but i make many time a simple conf for extensions.conf |
| 10:52.28 | WQDEQWDQW | i can make calls |
| 10:52.32 | WQDEQWDQW | but i cant receive |
| 10:52.49 | WQDEQWDQW | can you help me? |
| 10:54.05 | *** part/#asterisk WQDEQWDQW (n=weqwe@ip-37-175.sn1.eutelia.it) |
| 11:02.48 | *** join/#asterisk coppice_ (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 11:14.30 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 11:18.46 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 11:24.50 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 11:30.19 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 11:38.46 | *** join/#asterisk apardo (n=apardo@87.217.144.227) |
| 11:40.32 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 11:43.06 | DonX | Anyone have an example t1 config for a te110p as a t1? |
| 11:47.19 | dlynes_laptop | DonX: gimme a sec...just downloading the latest zaptel |
| 11:49.19 | dlynes_laptop | DonX: http://pastebin.ca/282218 |
| 11:50.43 | dlynes_laptop | DonX: that's using a sangoma u101, which is the same thing as far as zaptel is concerned |
| 11:51.06 | DonX | cool |
| 11:51.08 | DonX | thanks :) |
| 11:51.17 | dlynes_laptop | DonX: that was my zaptel.conf file |
| 11:51.24 | dlynes_laptop | DonX: did you want the zapata.conf file as well? |
| 11:51.33 | DonX | How do I tell it to do T-1 instead of PRI? |
| 11:51.42 | DonX | that would be nice too :) |
| 11:51.51 | dlynes_laptop | DonX: Also, the machine that that's in, is in a telecommunications colocation center |
| 11:52.07 | dlynes_laptop | DonX: so the lead-out is almost nil |
| 11:52.20 | DonX | would this be right for a t1? |
| 11:52.24 | DonX | span=1,1,0,esf,b8zs |
| 11:52.24 | DonX | bchan=1-24 |
| 11:52.25 | DonX | loadzone = us |
| 11:52.25 | DonX | defaultzone=us |
| 11:52.31 | dlynes_laptop | DonX: Well, I'm using a t1 pri |
| 11:52.39 | dlynes_laptop | DonX: i don't know about a straight t1 |
| 11:53.04 | dlynes_laptop | DonX: but yeah, i'm guessing just get rid of the dchan and replace it with a bchan |
| 11:53.07 | DonX | also, how do you target the extensions in extensions.conf |
| 11:53.14 | DonX | cool |
| 11:53.25 | dlynes_laptop | DonX: gimme a sec, and i'll post my zapata.conf file |
| 11:53.51 | dlynes_laptop | DonX: the extension naming in the dialplan only makes sense based on your particular zapata.conf file |
| 11:54.57 | dpenev | Hi I was sniffering the udp port 5060 on my local machine to see what is going wron with my configuration |
| 11:55.11 | dpenev | I have asterisk installed and several soft phones |
| 11:55.19 | DonX | alright |
| 11:55.31 | dlynes_laptop | DonX: http://pastebin.ca/282221 |
| 11:55.32 | dpenev | I see the following: |
| 11:55.32 | dpenev | U 2006/12/17 13:59:35.808105 127.0.0.1:5060 -> 127.0.0.1:5060 |
| 11:55.32 | dpenev | <PROTECTED> |
| 11:55.32 | dpenev | <PROTECTED> |
| 11:55.33 | dpenev | <PROTECTED> |
| 11:56.24 | DonX | hrmm |
| 11:56.47 | dlynes_laptop | DonX: Then you would use something like Dial(Zap/g1/6041234567) to dial |
| 11:57.41 | *** join/#asterisk dpenev (n=dpenev@89.190.200.195) |
| 11:57.54 | dlynes_laptop | dpenev: now you see what happens when you don't use pastebin |
| 11:57.56 | dlynes_laptop | ~pb |
| 11:58.02 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 11:58.23 | dpenev | paste bin? sorry I am new to IRC :-( |
| 11:58.35 | dlynes_laptop | dpenev: read jbot's message |
| 11:58.54 | dpenev | OK sorry guys! |
| 11:59.14 | dlynes_laptop | It's considered quite rude to post more than three lines to an irc channel at a time (2 lines in some channels) |
| 11:59.31 | dlynes_laptop | It's even more rude than typing in all CAPS LOCK |
| 12:00.01 | dpenev | Ok dlynes_laptop sorry again I did this because I didn't know that it is not good ... sorry |
| 12:00.15 | DonX | cool |
| 12:00.22 | dlynes_laptop | Yeah, it's cool |
| 12:00.24 | dlynes_laptop | don't worry about it |
| 12:00.33 | DonX | sorry aboutthat |
| 12:00.36 | dlynes_laptop | It's more of an issue when the channel's busy |
| 12:00.52 | dpenev | ok what should I do? |
| 12:01.09 | dlynes_laptop | Use pastebin |
| 12:01.14 | dpenev | ok it is clear |
| 12:01.21 | dlynes_laptop | To paste whatever it was you were trying to paste to the channel |
| 12:01.23 | dpenev | now can I ask my question? |
| 12:01.31 | dlynes_laptop | Sure |
| 12:02.03 | dpenev | I greap the udp 5060 trafic between softphone and * |
| 12:02.52 | dpenev | it seems to me that the users I put( dpenev and test) in sip.conf and extension.conf are not recognized |
| 12:03.21 | dpenev | I mena my softphone twinkle says that it can not register dpenev and test ... what can be the problem? |
| 12:03.31 | dlynes_laptop | dpenev: if you paste your asterisk log to pastebin, that'll be more useful to start with than a sip debug |
| 12:03.55 | dpenev | I set verbosity 10 and debug 10 at my * console but I don't see any life there? |
| 12:04.19 | dlynes_laptop | dpenev: check /var/log/asterisk/full (assuming you don't have it commented out in /etc/asterisk/logger.conf) |
| 12:04.21 | dpenev | wheer can I find this log? |
| 12:04.41 | dlynes_laptop | I just told you |
| 12:05.54 | DonX | what if I have ore than one group in zapata.conf? |
| 12:06.01 | DonX | IE: two cards |
| 12:06.27 | DonX | do I need to make two seperate contexts? |
| 12:06.38 | dpenev | well logger.conf was created by make samples as far as I undestood and I have not modified it |
| 12:06.42 | dlynes_laptop | Zap/g1/number for first group, Zap/g2/number for second group |
| 12:07.01 | DonX | yeah, but for the actual card |
| 12:07.04 | DonX | err |
| 12:07.05 | dlynes_laptop | dpenev: That doesn't preclude you from taking a look at it, to see if full is commented out or not |
| 12:07.10 | DonX | zapata.conf |
| 12:07.33 | dlynes_laptop | DonX: oh..no you don't need to make two separate contexts, if you don't want to |
| 12:07.36 | DonX | would I just do group => 2? Can that be in the same context? |
| 12:07.41 | DonX | cool |
| 12:07.45 | DonX | just define the diff chans |
| 12:07.47 | dlynes_laptop | DonX: just do bchan=1-48 instead in your zaptel.conf |
| 12:08.04 | DonX | well, they're going to be two seperate circuits |
| 12:08.09 | dlynes_laptop | DonX: and channel => 1-48 in your zapata.conf file |
| 12:08.27 | dpenev | I think it is commented, so I uncomment correct? |
| 12:08.27 | dpenev | ;full => notice,warning,error,debug,verbose |
| 12:08.34 | dlynes_laptop | DonX: two separate circuits in zaptel.conf? |
| 12:08.37 | DonX | I don't want to trunk them together or anything |
| 12:08.38 | DonX | yeah |
| 12:08.46 | DonX | one to one switch and another to the other one |
| 12:08.49 | dlynes_laptop | dpenev: remove the ';' in front of it, and then save it |
| 12:08.55 | dpenev | ok |
| 12:09.02 | dlynes_laptop | dpenev: then type 'logger reload' at the asterisk CLI |
| 12:09.09 | dlynes_laptop | dpenev: then reboot your phones |
| 12:09.17 | dlynes_laptop | dpenev: and pastebin the full file |
| 12:09.20 | dpenev | OK will do right now |
| 12:09.25 | dpenev | thansk |
| 12:09.51 | dlynes_laptop | DonX: I think you might still be able to set your zapata.conf so that it's channel => 1-48 if you want |
| 12:09.58 | dlynes_laptop | DonX: but I would try it first, myself |
| 12:10.14 | dlynes_laptop | DonX: I don't know off the top of my head whether it would work for sure, or not |
| 12:10.15 | DonX | well...there are going to e two seperate extensions ranges for each switch |
| 12:10.35 | dlynes_laptop | DonX: I don't follow you |
| 12:10.42 | dlynes_laptop | DonX: 'extension ranges'? |
| 12:10.51 | *** join/#asterisk alpinus (n=alpinus@swanky.hack.pl) |
| 12:10.51 | dlynes_laptop | DonX: is it analog phones hooked up to these t1's? |
| 12:10.57 | DonX | 6XXX goes to switch1 and 7XXX goes to switch two |
| 12:10.59 | dlynes_laptop | DonX: or co? |
| 12:11.08 | DonX | meridians |
| 12:11.10 | dlynes_laptop | DonX: ok, so it's analog phones, then |
| 12:11.37 | dlynes_laptop | DonX: you need to configure each channel individually in zapata.conf then |
| 12:11.55 | dlynes_laptop | DonX: iow, you won't be using channel groups |
| 12:12.01 | DonX | rmm |
| 12:12.21 | dlynes_laptop | DonX: each meridian only handles 24 extensions? |
| 12:12.25 | dpenev | strange I've uncomented full and for the first time I manage to logion to asterisk using xlite |
| 12:12.26 | DonX | I can't just config two groups? One for 1-24 and another for 25-48? |
| 12:12.35 | DonX | nah |
| 12:12.41 | dlynes_laptop | didn't think so |
| 12:12.46 | dlynes_laptop | I think you want pri, not t1, no? |
| 12:13.04 | dpenev | now aas I try to dial 4321 which is test user I see in * prompt '3 - No route to destination' |
| 12:13.05 | DonX | Yeah I'd like pri but there isn't a free d chan port on either switch |
| 12:13.11 | dlynes_laptop | It was my understanding that t1's can't tell what number you dialed other than by the channel you called in on |
| 12:13.20 | dpenev | any ideas? |
| 12:13.49 | dlynes_laptop | dpenev: you're not registered with asterisk |
| 12:13.53 | dlynes_laptop | dpenev: type sip show peers |
| 12:14.23 | dlynes_laptop | dpenev: you'll probably see a bunch of stuff in that display indicating how well it's not working |
| 12:14.48 | dpenev | I see 3 lines I will copy it here? |
| 12:14.51 | dlynes_laptop | sure |
| 12:15.03 | dpenev | so 3 lines are not so much :-) |
| 12:15.08 | dlynes_laptop | nah |
| 12:15.13 | dpenev | test/test (Unspecified) D 0 Unmonitored |
| 12:15.13 | dpenev | dpenev/dpenev 127.0.0.1 D 5061 Unmonitored |
| 12:15.17 | dlynes_laptop | 3 lines is the limit in this channel,a faik |
| 12:15.29 | dpenev | 2 sip peers [2 online , 0 offline] |
| 12:15.32 | dlynes_laptop | Ok, as you can see |
| 12:15.43 | dlynes_laptop | dpenev can make calls, no problem |
| 12:15.46 | dlynes_laptop | test cannot |
| 12:15.52 | dpenev | dlynes_laptop can ypou translate thease lines to me? |
| 12:15.55 | dlynes_laptop | well, actually |
| 12:16.00 | dlynes_laptop | test might be able to make calls |
| 12:16.05 | dlynes_laptop | but test won't be able to receive calls |
| 12:16.26 | dlynes_laptop | dpenev can probably make calls and can definitely receive calls |
| 12:16.27 | *** join/#asterisk puk_jp (n=ukris@p3143-ipad415marunouchi.tokyo.ocn.ne.jp) |
| 12:16.31 | dpenev | Ok what should I do to fix it? |
| 12:16.50 | dlynes_laptop | Check your log file to find out why test isn't registering |
| 12:17.23 | dlynes_laptop | I'm guessing you've got the wrong ip for your asterisk system saved in the 'test' phone, or you've got the wrong username and/or wrong password |
| 12:17.43 | dlynes_laptop | dpenev: You usually need to define a proxy and a registration server for each phone |
| 12:18.41 | dpenev | ok I will check again the settings of test at my softphone . thank you! but still it is not clear why everything become OK as I uncomented full in logger.conf? |
| 12:18.49 | dlynes_laptop | DonX: You'll probably be able to get more knowledgable help on the difference between regular t1's and pri-t1's in another 3 hours or so when the rest of north america starts to wake up |
| 12:18.59 | dlynes_laptop | dpenev: just coincidence |
| 12:19.21 | dpenev | coincidence ? with what? |
| 12:19.52 | dpenev | I played almost whole day to see some life in * prompt but it was necesery to uncomment full? |
| 12:20.21 | dpenev | I am pretty sure that it will stop working if I comment full again |
| 12:20.25 | dlynes_laptop | dpenev: what can I say? it probably just hates you |
| 12:20.29 | dlynes_laptop | dpenev: :) |
| 12:20.43 | dlynes_laptop | dpenev: but maybe it was the fact that you rebooted the phones |
| 12:20.57 | dlynes_laptop | dpenev: trust me...it had nothing to do with the log file being enabled |
| 12:22.38 | dpenev | yes I've just check that disabling full and still I can connect to *? strange things happens :-( |
| 12:22.56 | dpenev | I have reboot the phones probably 30 times |
| 12:23.13 | dlynes_laptop | but not after i told you to |
| 12:23.14 | dlynes_laptop | :) |
| 12:23.48 | dpenev | probably it is something with linux permisions... I came from windows world and I see that in linux the complicated file permissionsscheme often introduce problems , do you think it is posible? |
| 12:24.11 | dpenev | :-) Yes tahst right:-) |
| 12:24.11 | dlynes_laptop | dpenev: windows permissions are a lot more complicated than linux |
| 12:24.31 | dlynes_laptop | dpenev: the biggest difference is that most windows users don't use the permissions |
| 12:24.57 | dpenev | you mena NT well they both get the unix idea but it is fact that on linux I get lot of troubles unless I type su |
| 12:25.34 | dlynes_laptop | dpenev: same thing on windows if you don't have administrator privileges |
| 12:25.59 | dpenev | I mean strange behaviour which seems unrelated with file permissions at all but appears related |
| 12:32.02 | dpenev | now I see: sop show peers and I see ringing as I dial test from dpenev |
| 12:32.02 | dpenev | test/test 127.0.0.1 D 5061 Unmonitored |
| 12:32.02 | dpenev | dpenev/dpenev 127.0.0.1 D 5061 Unmonitored |
| 12:33.14 | dpenev | Can you tell pls. me now how I can test the system I have xlite and twinkle installed and one audio card on my PC I don't have analog phones and FXO/FXS for my PC |
| 12:33.40 | dpenev | is there voicemail or someting? |
| 12:34.40 | dlynes_laptop | dpenev: yes...get all your answers at the wiki |
| 12:34.42 | dlynes_laptop | ~wiki |
| 12:34.46 | dlynes_laptop | ~doc |
| 12:34.52 | jbot | Documentation can be found at http://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 |
| 12:34.52 | dlynes_laptop | ~docs |
| 12:34.54 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 12:35.04 | dlynes_laptop | ~thewiki |
| 12:35.06 | jbot | thewiki is, like, at http://www.voip-info.org/wiki-Asterisk |
| 12:35.45 | *** join/#asterisk RoyK (n=roy@ti211210a080-6347.bb.online.no) |
| 12:35.54 | dlynes_laptop | morning, roy |
| 12:36.05 | dpenev | Ok thank you all very much! I'll check the documentation now! |
| 12:38.06 | lenne_dk | Documentation is not always easy. This is from the doc of my ip-phone: "Enter setting mode: press this machine phone number key, newspaper finish copies of |
| 12:38.06 | lenne_dk | machine after the number first, the display screen appears : menu password: , input at this |
| 12:38.06 | lenne_dk | moment : 1234 passwords, press the handfree key; the display screen presents setting of |
| 12:38.06 | lenne_dk | # press, pin the # key and does not put at this time, see display screen |
| 12:38.06 | lenne_dk | appear"password:" ,Input: 1234, press the handfree key, the display screen appears : Network |
| 12:38.07 | lenne_dk | settings, the microphone enters the mode of setting up at this moment" |
| 12:38.20 | lenne_dk | Newspaper? :-) |
| 12:38.36 | masonc | GUI question: I have a plain vanilla asterisk install going in today to a lawyers office. The systems admin at the office is not capable of root access, but I would like to give him the ability to change the extensions info, especially email addresses for voicemail. The system is configured, I don't want to startt again with a freepbx install. What GUI would work for this? |
| 12:43.26 | *** join/#asterisk tuck3r_ (n=tuck3r@unaffiliated/tuck3r) |
| 12:44.50 | suma | hi |
| 12:45.40 | suma | i got a problem with iax calls |
| 12:45.45 | suma | anyone please help |
| 12:47.49 | shellshark | we cant help if you don't describe the problem |
| 12:48.05 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:48.57 | suma | when i make calls between iax to iax, it works fine, when i make calls between zap to iax the call drops |
| 12:49.13 | *** join/#asterisk stuq (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 12:49.18 | suma | zap is a pri line |
| 12:49.32 | qdk | suma: and you are use that it isnt zap? |
| 12:49.55 | suma | qdk: i did not get you |
| 12:49.56 | shellshark | qdk: s/use/sure/ ? |
| 12:50.07 | qdk | ups |
| 12:50.14 | qdk | suma: and you are sure that it isnt zap? |
| 12:50.17 | qdk | sorry. |
| 12:50.39 | suma | yes, i can make iax to zap and works fine |
| 12:50.42 | suma | it is not zap |
| 12:50.45 | suma | it is iax |
| 12:50.55 | suma | one moment let me get the basic debugging info |
| 12:51.25 | qdk | suma: does it work with a SIP channel? |
| 12:51.39 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:52.31 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 12:53.11 | suma | qdk: i'm sure it is problem with iax |
| 12:53.50 | suma | qdk: http://pastebin.ca/282270 |
| 12:53.57 | suma | this is the basic call flow |
| 12:55.21 | suma | qdk: point me which point you want more debug information, i will get you the same |
| 12:56.16 | suma | qdk: the problem is, the iax phone is ringing eventhough the call is dropped in the asterisk side, when i pick up the phone nothing happens |
| 12:56.38 | cpm | I have that happen |
| 12:58.18 | suma | cpm: you have solved that problem ? |
| 12:58.20 | qdk | suma: yes, it looks much like a problem with IAX: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock |
| 12:58.23 | tzafrir | masonc, make your own inteerface. It basically needs to rewrite voicemail.conf |
| 12:58.34 | tzafrir | Nothing more |
| 12:59.09 | qdk | suma: I have never used IAX for anything else than trunks, and I dont even use that anymore. (or at least im phasing them out). |
| 12:59.15 | cpm | suma, naw, I don't worry about it. It happens to me, when iax2 extensions call some extensions on my channel bank, and I'm not sure my channel bank is properly detecting the hangup |
| 12:59.32 | cpm | eventually it stops, but way way past when it should |
| 12:59.55 | DonX | Does anyone have a plan T1 (non-pri) working with asterisk? |
| 13:00.18 | qdk | DonX: plan = plain? |
| 13:00.28 | qdk | DonX: answer still no though. :-) |
| 13:00.52 | dlynes_laptop | qdk: zapata.conf/zaptel.conf/extensions.conf for a t1 as opposed to a pri t1 |
| 13:01.15 | qdk | dlynes_laptop: huh? |
| 13:01.15 | dlynes_laptop | qdk: He's trying to hook some meridians up to asterisk via t1 links (w/o pri signalling) |
| 13:01.21 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.156.81) |
| 13:01.29 | dlynes_laptop | qdk: so that he can dial the extensions directly |
| 13:01.40 | dlynes_laptop | qdk: he has more than 24 extensions on each meridian |
| 13:02.09 | dlynes_laptop | qdk: i'm referring to DonX's issue |
| 13:03.02 | qdk | dlynes_laptop: well he asked for people with plan (which i guess is a plain) T1 working with *, and I just replied no... Really dont care whatever he is using it for. ;-) |
| 13:03.39 | DonX | yeah sorry |
| 13:04.10 | DonX | hehe |
| 13:04.58 | Druken | anyone know where i could find a listing of early media messages of telephone companies? |
| 13:12.58 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:13.32 | pif | hi, I get this regularly at the * console: "midget packet received (1 of 4 min)" |
| 13:13.40 | pif | how can I get rid of it? |
| 13:16.50 | dlynes_laptop | pif: It might be getting caused because your network is having issues if you're getting a lot of them |
| 13:18.26 | dlynes_laptop | pif: it's an iax error, usually |
| 13:18.27 | pif | I get it every 30 seconds |
| 13:18.39 | pif | precisely |
| 13:18.42 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 13:19.29 | dlynes_laptop | pif: is the iax communication happening over the local area network? |
| 13:19.29 | dlynes_laptop | pif: or is there some distance communication involved? |
| 13:19.38 | pif | distant hosts are involved |
| 13:20.01 | *** join/#asterisk isladelobos (n=jjj@43.Red-83-38-108.dynamicIP.rima-tde.net) |
| 13:20.05 | dlynes_laptop | pif: check to make sure you're not getting major packet loss on your external ip address |
| 13:20.35 | pif | it happens on all hosts involved |
| 13:20.46 | dlynes_laptop | pif: normally that's just a warning, but if you're getting it repeatedly, i would suspect there's a bigger problem |
| 13:20.56 | dlynes_laptop | pif: Yes, but you only have one external ip address, right? |
| 13:21.04 | pif | yes |
| 13:21.16 | dlynes_laptop | pif: yeah...check for packet loss on it, if you can |
| 13:21.20 | pif | ok, i'll dig deeper, thanks |
| 13:21.27 | dlynes_laptop | pif: are you using wireless to get to the gateway? |
| 13:21.47 | pif | no wireless, connexions are professional grade |
| 13:21.50 | dlynes_laptop | pif: ok |
| 13:22.02 | dlynes_laptop | pif: anyways...from what i can see by digging into the code |
| 13:22.20 | dlynes_laptop | pif: that error is caused because it was only able to get a partial packet where it was expecting a full packet |
| 13:22.43 | pif | the firewall might be involved... |
| 13:22.44 | dlynes_laptop | That's why I suspect it might be an issue with some packet loss on the network somewhere |
| 13:23.51 | *** join/#asterisk justin__ (n=justin@5ac0320c.bb.sky.com) |
| 13:24.31 | pif | what's really vexing is google returns nothing on this, I'm alone :-/ |
| 13:25.15 | *** join/#asterisk lanec (i=lanec@bainbrdg-cuda1-69-161-211-36.clvdoh.adelphia.net) |
| 13:25.19 | lanec | Hello |
| 13:25.40 | lanec | is there anyone here who can help answer some questions I have about Asterisk? |
| 13:26.43 | pif | don't ask to ask |
| 13:27.05 | lanec | haha just making sure someone was on |
| 13:27.10 | lanec | if no one responded I would disconnect |
| 13:27.38 | lanec | Is it possible to forward say 4 voip lines to an asterisk box and have the asterisk box manage them? |
| 13:27.43 | dlynes_laptop | pif: try using ethereal, ettercap, ngrep, sipsak, sipp |
| 13:28.01 | dlynes_laptop | pif: well, actually sipsak and sipp aren't going ot help you with that issue |
| 13:28.04 | pif | the message is from chan_iax2.c |
| 13:28.06 | lanec | I know you can take 4 land lines and share them over the web |
| 13:28.18 | dlynes_laptop | pif: but ethereal should be able to help for sure |
| 13:28.27 | pif | good idea |
| 13:28.39 | dlynes_laptop | pif: who knows...the problem may even be a bug in the iax2 channel code |
| 13:28.50 | dlynes_laptop | pif: but i suspect something on your network is the issue, not the code |
| 13:29.16 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 13:29.46 | dlynes_laptop | pif: but at least if ethereal is showing a full iax2 packet, and asterisk is reporting an issue, then you'll know it's an issue in the channel driver |
| 13:30.25 | dlynes_laptop | pif: or perhaps ethereal will show an incomplete packet, and then you'll know it's a network issue, or perhaps it's an iax2 issue on the remote end |
| 13:30.48 | dlynes_laptop | anyways...i need sleep |
| 13:30.50 | dlynes_laptop | good night |
| 13:31.20 | lanec | Good night |
| 13:32.17 | lanec | I am sorry if my questions are lacking obvious technical specifications such as protocols or bandwidth needs but I am asking in extreme basics here |
| 13:32.36 | lanec | can Asterisk support 4 incoming voip lines through something like sips and turn them into copper lines? |
| 13:32.45 | lanec | while handling voicemail and call forwarding for them |
| 13:33.13 | lanec | if the answer is yes I plan to explore the documentation much more |
| 13:33.18 | lanec | if not I would like to save myself some time |
| 13:33.59 | Druken | what do you mean turn them into copper lines? |
| 13:34.05 | dlynes_laptop | lanec: the short answer is yes |
| 13:34.06 | pif | dlynes_laptop : good night |
| 13:34.39 | lanec | okay, is this something that can be done without writing my own scripts? such as through a gui or tutorial on the subject? |
| 13:34.53 | lanec | or following a tutorial on the subject* |
| 13:35.28 | Druken | uhm... doing anything with asterisk requires some intelligence, and brain work... |
| 13:35.47 | Druken | asterisk is not for the stupid and lazy... ok well... maybe for the lazy.. but not the stupid |
| 13:36.26 | lanec | I mean I have standard phone lines in a small business I do some work for |
| 13:36.29 | pif | learning curve is very steep, consider hiring a consultant |
| 13:36.38 | lanec | they have been getting cut off by long distance providers for making to many phone calls |
| 13:36.45 | pif | unless your time is worthless |
| 13:36.47 | lanec | I have lots of experience with Linux |
| 13:36.52 | lanec | and network security |
| 13:36.54 | Druken | pif: steep? it's like a hairturn turn.... |
| 13:37.02 | lanec | well I won't know untill I try |
| 13:37.15 | pif | hairpin? |
| 13:37.16 | Druken | er.. hairpin turn |
| 13:37.16 | lanec | frankly this isn't exactly a warming experience |
| 13:37.17 | Druken | ehehe |
| 13:37.21 | lanec | lol |
| 13:37.27 | lanec | although i'm sure its necessary |
| 13:37.52 | lanec | I am downloading asterisk now at this time and I will give it a spin in an hour or so |
| 13:38.05 | Druken | hehehe we tend to be a bunch of crusty bastards... especially early mornings |
| 13:38.20 | lanec | anyone going to object to me coming back here for more of these questions? or am I just annoying you grumps ;-D |
| 13:38.45 | pif | we suffered to master asterisk |
| 13:38.54 | Druken | do some reading while your waiting :) |
| 13:39.03 | pif | so you should sweat a little :) |
| 13:39.03 | Druken | ~docs |
| 13:39.30 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 13:39.30 | lanec | yeah sounds good |
| 13:39.30 | Druken | hmm.... jbot broken ? |
| 13:39.33 | lanec | thanks guys i'm sure i'll be back later after I get a bit of this under my belt |
| 13:39.40 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 13:39.42 | Druken | ahh, just slow this morning like everyone else |
| 13:40.02 | EmleyMoor | When I finish a call, I get this: |
| 13:40.17 | EmleyMoor | WARNING[7279]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 13:40.18 | EmleyMoor | <PROTECTED> |
| 13:40.23 | EmleyMoor | Why would that be? |
| 13:40.30 | Druken | check your dialplan |
| 13:40.37 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 13:40.48 | Druken | are you doing some kind of math after you hangup ? |
| 13:41.00 | EmleyMoor | No |
| 13:41.15 | EmleyMoor | I do nothing special on hangup unless the call is an emergency call |
| 13:41.20 | pif | obsolete bison libraries? |
| 13:41.27 | pif | or yack? |
| 13:41.47 | EmleyMoor | I suppose that's vaguely possible |
| 13:42.02 | pif | all _yy functions use bison/yack |
| 13:42.09 | dlynes_laptop | EmleyMoor: it's a bug in your dialplan....probably a space where one doesn't belong, or something |
| 13:42.47 | EmleyMoor | Any clue as to what it is near? |
| 13:42.56 | dlynes_laptop | EmleyMoor: no idea without seeing your dialplan |
| 13:43.06 | EmleyMoor | It could well be outdated bison as pif suggests |
| 13:43.11 | Druken | i would guess near the hangup ? |
| 13:43.11 | dlynes_laptop | EmleyMoor: and I just forgot something i was doing, or i'd be sleeping already |
| 13:43.32 | dlynes_laptop | EmleyMoor: doubt it's an outdated bison...it's a possibility, but doubtful |
| 13:43.52 | EmleyMoor | There is no specific hangup on this kind of call |
| 13:44.31 | dlynes_laptop | EmleyMoor: just pastebin your dialplan where you're having this error |
| 13:48.03 | dlynes_laptop | EmleyMoor: have you pastebinned it yet? |
| 13:48.18 | EmleyMoor | No |
| 13:50.14 | dlynes_laptop | Just pushing you because I need to get to sleep, and that other job I forgot I was doing is finished now |
| 13:50.17 | EmleyMoor | http://pastebin.ca/282311 |
| 13:51.27 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:51.49 | dlynes_laptop | EmleyMoor: What's ${BT}? |
| 13:52.05 | dlynes_laptop | EmleyMoor: and what's ${VT}? |
| 13:52.33 | EmleyMoor | Zap/4 and IAX2/xxxxxxxx@voiptalk (xs representing numbers) |
| 13:52.58 | dlynes_laptop | so BT is Zap/4? |
| 13:53.06 | dlynes_laptop | and VT is IAX2/xxxxxxxx@voiptalk? |
| 13:53.16 | EmleyMoor | Yes |
| 13:53.37 | dlynes_laptop | EmleyMoor: Then it's your third line that's not making any sense |
| 13:53.47 | EmleyMoor | ... why?# |
| 13:53.58 | *** part/#asterisk isladelobos (n=jjj@43.Red-83-38-108.dynamicIP.rima-tde.net) |
| 13:53.59 | dlynes_laptop | It becomes Dial(IAX2/xxxxxxxxx@voiptalk/44xxxxxxxxxxx) |
| 13:54.07 | *** join/#asterisk isladelobos (n=jjj@43.Red-83-38-108.dynamicIP.rima-tde.net) |
| 13:54.20 | EmleyMoor | Yes... that is correct |
| 13:54.21 | dlynes_laptop | After all the expansion kicks in |
| 13:54.32 | dlynes_laptop | You've got your number in there twice |
| 13:54.41 | EmleyMoor | No, not the same number |
| 13:54.56 | Druken | it's got his username in the variable |
| 13:54.57 | dlynes_laptop | Ok, so $VT is the iax with the username |
| 13:55.03 | EmleyMoor | Yes |
| 13:55.18 | dlynes_laptop | Yeah...I don't see anything wrong with it then |
| 13:55.31 | dlynes_laptop | Unless your problem is somewhere else |
| 13:55.45 | dlynes_laptop | To know where it is for sure |
| 13:55.56 | dlynes_laptop | You can add in a bunch of Noops |
| 13:56.31 | dlynes_laptop | And then you'll know which line of your dialplan, specifically is causing it |
| 13:56.42 | dlynes_laptop | anyways...on that note, i'm going to sleep |
| 13:56.43 | EmleyMoor | No new bison anyway |
| 13:56.44 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 13:56.45 | dlynes_laptop | good night, again |
| 14:01.39 | lenne_dk | Queues: I've got two phones listed in a queue, the second with a penalty. If the first phone is busy, the second is called when a second caller is in the queue. But if the first just doesn't answer, the second is never called. How can I make the call roll over to the second phone if either the first is busy (as it does now) or when the first doesn't answer (which it doesn't now) |
| 14:02.20 | mosty | set a timeout on the first queue member |
| 14:02.33 | lenne_dk | how? |
| 14:02.50 | mosty | i think it's a field of the queue settings |
| 14:03.02 | mosty | not on the member (sorry), i meant the queue |
| 14:03.18 | mosty | that, or limit the number of simultaneous calls to the sip user in sip.conf |
| 14:04.29 | lenne_dk | I don't see how the simultaneous calls can help. The first caller is still ringing on the first phone forever. |
| 14:05.42 | mosty | that would fix another common queue problem, where the first queue member has taken a call and gets sent another before they're finished |
| 14:06.58 | Druken | very annoying when that happens :) |
| 14:08.04 | mosty | Queue sucks :/ |
| 14:08.20 | lenne_dk | 36949608 has 2 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s |
| 14:08.20 | lenne_dk | <PROTECTED> |
| 14:08.20 | lenne_dk | <PROTECTED> |
| 14:08.20 | lenne_dk | <PROTECTED> |
| 14:08.20 | lenne_dk | <PROTECTED> |
| 14:08.21 | lenne_dk | <PROTECTED> |
| 14:08.23 | lenne_dk | <PROTECTED> |
| 14:08.27 | Druken | ~pastebin |
| 14:08.29 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:10.03 | pif | dlynes_laptop : found what caused the midget packet |
| 14:10.31 | pif | it's a monitoring program that sends probes, wmnetmon, thanks for you suggestions |
| 14:11.12 | pif | on what port do you guys monitor an active asterisk? |
| 14:11.20 | *** join/#asterisk mustafa (n=mustafa@202.141.252.73) |
| 14:12.12 | lenne_dk | Perhaps I should use 36949608 roundrobin: take turns ringing each available interface. |
| 14:12.17 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 14:12.26 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 14:13.14 | lenne_dk | But I can just imagine some poor guy alone at the office chasing arould trying to catch the call as it goes from one phone to the next :-) |
| 14:14.00 | mosty | lenne_dk, use rrmemory instead of roundrobin |
| 14:14.59 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 14:16.06 | lenne_dk | No, the first phone should always be tried first, only if not avalilable/busy/not answering should the second be called. rrmemory will call the first and second on alternative callers, I think |
| 14:16.10 | *** join/#asterisk mustafa (n=mustafa@202.141.252.73) |
| 14:16.31 | mustafa | hi |
| 14:17.06 | mustafa | how can i get actuall call duration? |
| 14:17.31 | lenne_dk | But I have two cabled phones and two wireless phones on my desk, so I can easily find out... |
| 14:17.58 | mustafa | i mean bridged call duration |
| 14:18.47 | lenne_dk | the duration is available in the CDR log. |
| 14:19.04 | mustafa | its not the actuall call duration |
| 14:19.06 | mustafa | like |
| 14:19.12 | mosty | lenne_dk, doesn't the penalty take precedence over the round robin-ness? not really sure about that |
| 14:19.41 | mosty | lenne_dk: but if you only have two queue members, you probably don't need a queue, you could write your own dialplan code for that |
| 14:19.44 | mustafa | it adds 5 to 10 extra seconds in call time |
| 14:20.15 | mosty | mustafa, duration and billsec don't give you what you want? |
| 14:20.33 | mustafa | forexample i dial a cellfone number from my ip phone. it takes me to asterisk server. where i have some agi scripts taking 5,10 seconds |
| 14:21.27 | mustafa | plus for the time other party is ringing my cdr also includes that time |
| 14:21.39 | mustafa | what i want is actuall Bridged call duration |
| 14:22.01 | *** join/#asterisk Skarmeth (n=Skarmeth@201009058078.user.veloxzone.com.br) |
| 14:22.03 | *** join/#asterisk brif8 (n=brif8@rrcs-67-78-24-179.se.biz.rr.com) |
| 14:22.10 | mosty | can you call the agi script before Answer'ing the call on the asterisk server? |
| 14:22.41 | mustafa | i have no idea if its possible. |
| 14:22.42 | Druken | mustafa: that is what billsec is for |
| 14:22.51 | mustafa | i dont knwo about it |
| 14:22.53 | mustafa | let me see |
| 14:23.11 | Druken | in cdr you have duration, and billsec, billsec is the billable seconds |
| 14:23.19 | Druken | from ANSWER to HANGUP |
| 14:23.25 | *** part/#asterisk brif8 (n=brif8@rrcs-67-78-24-179.se.biz.rr.com) |
| 14:23.31 | mustafa | oh i c |
| 14:23.59 | mustafa | excellent! |
| 14:27.00 | lenne_dk | queues suck :-) |
| 14:27.23 | Druken | queues are wonderfull things.... |
| 14:27.27 | Druken | once setup properly |
| 14:27.31 | mosty | lenne_dk, akkurat |
| 14:28.44 | mosty | well, queues are wonderful things, but Queue sucks ;) |
| 14:28.45 | mustafa | thanks |
| 14:29.31 | lenne_dk | I tried rrobin, no penalties, and two inbound calls. With nobody answering, only one phone was ringing at a time. I'd still like the fancy stuff with MOH and "you are number 27 in the queue", but I probably need to do it in dialplan instead... |
| 14:30.00 | Druken | you want them both to ring? |
| 14:30.07 | Druken | then use ringall |
| 14:30.39 | mosty | lenne_dk, rrobin only appears useful if the queue is never empty |
| 14:30.50 | *** join/#asterisk ucfMethod (n=ucfmetho@c-69-143-112-70.hsd1.va.comcast.net) |
| 14:31.14 | lenne_dk | But phone 2 should not ring if phone 1 is available. |
| 14:31.45 | Druken | so use priorities |
| 14:32.00 | lenne_dk | Perhaps I should first call phone 1 with a short timeout, and then if not answered, put it in the queue. |
| 14:33.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:33.10 | lenne_dk | But I'd like it announced if it is a call from the private line or the office line, as queue-announce can. |
| 14:33.43 | mosty | lenne_dk: i use caller id for that |
| 14:34.15 | lenne_dk | But the phones doesn't all support that... |
| 14:34.30 | Druken | icky.... |
| 14:34.59 | mosty | i use standardised equipment for queues |
| 14:36.41 | pif | what is the "standard" dial code to suppress caller id? |
| 14:37.06 | pif | what is generaly used? |
| 14:37.42 | Druken | dial without passing CID ? |
| 14:38.12 | pif | yes |
| 14:39.11 | nibbler_de | has anybody of you clue how i can play music to the caller without using the music-on-hold feature? i'd like to start with the audio-file the caller hears when the application is called via the extensions.conf |
| 14:39.51 | Druken | pif: i belive you have to set it to nothing before dialing |
| 14:40.19 | Druken | nibbler_de: playback ? |
| 14:40.21 | pif | but then the telco display your default |
| 14:40.42 | Druken | pif: try "private" <> |
| 14:41.21 | Druken | or ask your telco how to eliminate the cid, what do you need to set it as to have it come up as private |
| 14:41.29 | nibbler_de | Druken: i want to have the audio-file while the destination party is ringing |
| 14:41.58 | Druken | no idea if you can without moh... |
| 14:42.07 | Druken | why not just use moh? |
| 14:42.29 | *** join/#asterisk af_ (n=af@ip-131-134.sn2.eutelia.it) |
| 14:42.30 | lenne_dk | m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. |
| 14:42.42 | lenne_dk | an option to Dial() |
| 14:42.53 | Druken | he wanted to do it WITHOUT moh |
| 14:43.49 | lenne_dk | Ok, then give us the reason for not using MOH to do MOH. |
| 14:43.51 | *** part/#asterisk dpenev (n=dpenev@89.190.200.195) |
| 14:44.42 | nibbler_de | Druken: i have 120 seconds of music here - the first 10 seconds are like an intro, then the call is forwarded to the callcenter, after 90 seconds there is an outro of 20 seconds that asks you to leave a message. the 120 seconds music in the background should play seamlessly |
| 14:46.18 | Druken | ok, so you want to answer, play 10 seconds of audio... ring an extension for 190 seconds, and play a 20 second voicemail into |
| 14:46.22 | Druken | sound about right? |
| 14:46.56 | nibbler_de | yup |
| 14:47.10 | lenne_dk | nibbler: perhaps you can "premix" the audio and the music externally of asterisk? |
| 14:47.27 | nibbler_de | lenne_dk: this has already happened |
| 14:48.21 | lenne_dk | So why bother finding another solution? :-) |
| 14:49.07 | nibbler_de | the problem is that asterisk will loop the 90sec part and play it to the caller from an arbitrary position when i use the m flag |
| 14:49.27 | nibbler_de | so the music and the announcements aren't seamlessly connected anymore |
| 14:49.50 | *** join/#asterisk Tili (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net) |
| 14:50.13 | nibbler_de | i want asterisk to start playing the 90sec of 'wait' for every caller from beginning |
| 14:50.39 | Druken | i wonder if you could background the file, and dial the extension |
| 14:50.52 | nibbler_de | "background the file"? |
| 14:51.03 | Druken | background(audio.gsm) |
| 14:51.07 | nibbler_de | hmm |
| 14:51.20 | Druken | i doubt that would work... but i guess it'd be worth a try |
| 14:55.10 | *** join/#asterisk humbolto (n=elias@u-121-071.adsl.univie.ac.at) |
| 14:55.23 | nibbler_de | doesn't work :( |
| 14:55.52 | humbolto | what do I need to connect an asterisk server to a dms100? |
| 14:57.14 | mosty | what's a dms100? |
| 14:57.36 | humbolto | telephony switch |
| 14:57.40 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:58.17 | puzzled | hi |
| 14:58.55 | mosty | humbolto, analogue? use a digium tdm card, or similar |
| 14:59.22 | benjk | humbolto a PRI card |
| 15:00.11 | humbolto | can I do that with the free software version or do I need an enterprise edition for that? |
| 15:00.35 | mosty | the free software version is fine |
| 15:02.14 | humbolto | when do I need the enterprise edition? |
| 15:02.46 | lenne_dk | ENUMLOOKUP: exten => 551,1,Set(foo=${ENUMLOOKUP(+4536949608,sip,,e164.org)}) |
| 15:02.46 | lenne_dk | exten => 551,2,noop(${ENUMSTATUS}) |
| 15:02.47 | lenne_dk | exten => 551,3,noop(${ENUM}) |
| 15:03.02 | lenne_dk | Why is ENUMSTATUS not set? |
| 15:03.28 | lenne_dk | log says : |
| 15:03.29 | benjk | humbolto, http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems |
| 15:03.35 | lenne_dk | -- Executing Set("SIP/ip-085f0000", "foo=leif@arnold.neland.dk") in new stack |
| 15:03.35 | lenne_dk | <PROTECTED> |
| 15:03.47 | benjk | T1 cards will do |
| 15:03.53 | lenne_dk | so enumlookup is working |
| 15:04.16 | humbolto | I would like quickly setup an asterisk testing environment. nothing fancy, just to test the voip part of the system. are there any nice distros out there? |
| 15:04.20 | benjk | humbolto, entreprise edition is never needed, under no circumstances |
| 15:04.21 | mosty | humbolto, get the enterprise edition if you want printed manuals an support, amongst otherthings |
| 15:04.36 | benjk | 20 hours of support, you can buy that separately |
| 15:05.06 | lenne_dk | Enterprise solution is for when the boss thinks "if it's free, it's worthless" |
| 15:05.06 | benjk | the entreprise edition is proprietary closed source software |
| 15:05.06 | *** part/#asterisk [Airwolf] (n=airwolf@89.205.156.81) |
| 15:05.15 | mosty | i think you're best off with the free software version, on your preferred dist, and pay for support by itself (if you need it) |
| 15:05.22 | benjk | and you get the same vendor lock-in that you get with any other proprietary PBX |
| 15:05.29 | humbolto | benjk: I understand. But I thought you need the enterprise edition for special environments where special codecs are needed. |
| 15:05.36 | benjk | no |
| 15:05.39 | humbolto | when you use certain hardware |
| 15:05.46 | mosty | humbolto, no |
| 15:06.29 | pif | humbolto : you need the Extreme edition |
| 15:06.56 | humbolto | pif: extreme edition? |
| 15:07.11 | humbolto | do you know any good voip upstream provider, I could hook my asterisk server up to? |
| 15:07.34 | mosty | there are lots. look at the wiki for providers in your area |
| 15:11.16 | lenne_dk | You can also look for providers, which have (free) numbers in areas, where you want to appear to be. I have just setup a number in London. |
| 15:11.25 | lenne_dk | And I'm in denmark |
| 15:11.26 | humbolto | do you know a version which runs on my wrt54gl? |
| 15:11.34 | humbolto | linksys wlan router |
| 15:11.56 | humbolto | lenne_dk: london sounds good |
| 15:12.01 | benjk | humbolto, will not interface with the DMS100 |
| 15:12.11 | lenne_dk | show version |
| 15:12.30 | humbolto | lenne_dk: and I find links to these providers at the asterisk wiki? |
| 15:12.40 | mosty | humbolto, the wrt54gl does not have the right hardware |
| 15:13.03 | benjk | asterisk wiki? whats that? |
| 15:13.13 | humbolto | mosty: what, to interface with a dms100, certainly |
| 15:13.18 | humbolto | no, just for home use |
| 15:13.19 | benjk | do you mean Voip-Info.org ? that's not an Asterisk wiki |
| 15:13.45 | benjk | to interface with a DMS100 you need T1 |
| 15:13.52 | lenne_dk | Pure luck :-) I first got a +44 871 number, but as it is an overcharged number, it can't be used from cellphones and some voip isp's |
| 15:16.53 | humbolto | what hardware requirements do I have for a pure voip asterisk server? |
| 15:16.54 | *** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net) |
| 15:17.01 | humbolto | for 1-30 users |
| 15:17.19 | humbolto | no, for max 10 concurrent connections |
| 15:17.20 | benjk | users means nothing |
| 15:17.34 | humbolto | right |
| 15:17.34 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 15:17.38 | TripleFFFF | anyone use mediatrix ? |
| 15:17.43 | benjk | the linksys is too weak for that |
| 15:17.55 | benjk | maybe a Soekris will do |
| 15:18.11 | benjk | or a WRAP |
| 15:18.23 | mosty | humbolto: depends what codec you use, and if the pbx will be transcoding |
| 15:18.37 | TripleFFFF | 10http://www.mediatrix.com/products_gateways.php?prodid=801 |
| 15:18.42 | TripleFFFF | is this worth anything |
| 15:20.53 | TripleFFFF | http://www.mediatrix.com/products_gateways.php?prodid=8 |
| 15:20.54 | benjk | 10 concurrent SIP or IAX calls with alaw/ulaw you could probably handle on this: http://www.pcengines.ch/wrap.htm or this: http://www.soekris.com/net4801.htm |
| 15:20.55 | TripleFFFF | ? |
| 15:22.43 | mosty | i have a net4801, it's about as fast as a pentium 133 or so |
| 15:23.59 | benjk | actually the Geode SC1001 is a Pentium 266 |
| 15:24.05 | TripleFFFF | i guess not then |
| 15:24.18 | Druken | TripleFFFF: my god man... what's with the bold? |
| 15:24.23 | benjk | well, equivalent to it |
| 15:24.59 | mosty | benjk, it feels like my p133, i've never used a p2-266 to compare with |
| 15:25.03 | Druken | TripleFFFF: i personally have no used mediatrix, however, i have not heard anything bad about them either |
| 15:25.18 | Druken | i belive thinktel out of edmonton uses them often |
| 15:25.29 | benjk | the AMD chip on those is equivalent to a P266 |
| 15:25.31 | mosty | benjk, it's 266Mhz sure, but it's not clock for clock equivalent to a pentium |
| 15:25.50 | benjk | next year they launch a new board with the newer 500MHz Geode |
| 15:25.55 | benjk | about March or so |
| 15:26.10 | lenne_dk | Is PSTN = POTS or PSTN = ISDN + POTS, i.e. is POTS = analog |
| 15:26.30 | Druken | pstn == publicly switched telephone network |
| 15:26.33 | benjk | POTS is analog, ISDN is digital |
| 15:26.34 | lenne_dk | ISDN != PSTN? |
| 15:26.37 | mosty | it's a great little router, too slow for anything complex though |
| 15:26.44 | benjk | ISDN is one part of the PSTN |
| 15:27.19 | lenne_dk | so PSTN = ISDN + POTS |
| 15:27.20 | benjk | mosty, it is a P1-266 |
| 15:27.26 | benjk | not a PII |
| 15:27.38 | Druken | lenne_dk: POTS == plain old telephone service |
| 15:27.55 | benjk | ISDN and POTS are the two last mile technologies of the PSTN |
| 15:27.56 | Druken | what ma'bell serves to residential customers |
| 15:28.00 | lenne_dk | is mobile phones also considered PSTN? |
| 15:28.11 | benjk | the core network of the PSTN runs on SS7 |
| 15:28.15 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 15:28.16 | mosty | benjk, in any case, it's no speed demon |
| 15:28.26 | *** join/#asterisk humbolto (n=elias@u-121-071.adsl.univie.ac.at) |
| 15:28.38 | benjk | no, but it should be able to handle about 10 ulaw/alaw calls |
| 15:29.53 | benjk | and the new boards due out in March might be able to handle a full T1 without transcoding |
| 15:30.28 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:31.39 | EmleyMoor | Is there a way to make my Zap phone give a special ring cadence based on the number that was called? |
| 15:33.13 | EmleyMoor | (I am sure there is - probably pointing me to an example would be enough - will search though) |
| 15:36.46 | xheliox | I have some sort of PLX T400P clone, with a PCI ID of 0x3000, looking at tor2-hw.h, I don't see that PCI ID. I tried just adding it using one of the other Tor2 lines as a template, but that didn't seem to work. At least it doesn't load the card. Is anyone familar with these pieces of junk and how to get them detected? :) |
| 15:36.56 | *** part/#asterisk KenSentMe (n=KenSentM@a82-92-80-8.adsl.xs4all.nl) |
| 15:37.21 | xheliox | Dlsclaimer: I had no part in deciding to buy this card. It was dropped in my lap. |
| 15:39.29 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 15:43.51 | mosty | how much time are you going to spend on this clone vs how much would a name brand card cost? |
| 15:44.21 | xheliox | mosty: Don't lecture me. I've already given the lectures. |
| 15:44.59 | EmleyMoor | Hmmm... default dring here is 0,0,0 |
| 15:47.09 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 15:47.27 | benjk | tormenta-2 cards are by default all clones |
| 15:48.10 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 15:48.44 | benjk | its GPL hardware, designed by Jim Dixon and put up for download for the sole purpose of being "cloned" |
| 15:48.57 | xheliox | Ok, fair enough. |
| 15:49.21 | EmleyMoor | Anyone in the UK with a BT line? (not necessarily on asterisk) |
| 15:50.02 | benjk | if you have trouble it could be a result of recent legislation in Europe which bans certain toxic substances in electronic circuit boards |
| 15:50.15 | benjk | also known as ROHS |
| 15:50.39 | benjk | some of the card manufacturers have altered the design to use parts which are ROHS compliant |
| 15:50.49 | benjk | this may require patches to the driver |
| 15:51.20 | tzafrir | But the fun part is that code from driver of those cards does not seem to be merged back into the Zapata project. |
| 15:51.50 | xheliox | My biggest problem is, I don't even know who manufactured it. |
| 15:51.58 | EmleyMoor | Other than in zapata.conf, do I have to make any settings to make dring detection work? |
| 15:51.59 | xheliox | The guy who sold it too him lied and said it was a Digium. |
| 15:52.04 | benjk | I know of one card which was modified for ROHS compiance and the manufacturer did submit the patches |
| 15:52.18 | tzafrir | EmleyMoor, not that I know of. |
| 15:52.20 | xheliox | Would that be in Mantis? |
| 15:52.38 | EmleyMoor | OK - so I just need to find someone with a BT line to help me test... |
| 15:52.46 | benjk | xheliox, that consitutes fraud |
| 15:53.09 | xheliox | benjk: And how is that my problem now? |
| 15:53.10 | *** join/#asterisk Menace- (i=menace@66.181.104.13) |
| 15:53.19 | benjk | if you have any name or contact details for this guy, you can report it to the local authorities |
| 15:53.46 | xheliox | yeah, yeah.. I'll let him worry about that, I'm just concerned about getting his phone system online |
| 15:54.17 | xheliox | He's already ordered a true blue Digium, but it won't be there until Tuesday morning and that's an entire business day without a phone system. |
| 15:54.26 | benjk | well, you could start by verifying the components on the card against the BoM on zapatatelephony.org |
| 15:54.28 | *** join/#asterisk BitBandit (n=polx@24-241-129-98.static.stgr.ut.charter.com) |
| 15:54.56 | benjk | if some parts are different, then it is probably modified for ROHS compliance and then it probably needs patched drivers |
| 15:55.19 | benjk | if the parts are all the same, then you know you have a good chance to get it working |
| 15:55.36 | xheliox | I don't care about having to patch code if I can find the code. :) |
| 15:55.56 | benjk | finding the code is difficult if yo don't know who made the card |
| 15:56.20 | benjk | in any event, the first step should be to compare the parts with the BoM to see if it actually has been modified |
| 15:56.56 | *** join/#asterisk Rhiliam (n=gary@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 15:57.21 | Rhiliam | is there way to group sip channels the way you group POTS channels? |
| 15:59.23 | xheliox | benjk: I've actually done that, they good very similar.. |
| 15:59.50 | *** join/#asterisk Winkie (n=urmom@host86-130-188-100.range86-130.btcentralplus.com) |
| 15:59.50 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:59.50 | benjk | very similar isn't good enough, you need to verify if the components are the same as on the BoM |
| 16:00.28 | Druken | does anyone have allison saying i'm sorry? |
| 16:00.47 | benjk | I seem to remember the most likely one to be different for ROHS is some MAXIM RS422 chip |
| 16:01.11 | benjk | maybe even the PCI bridge, but its been a while since I looked at that |
| 16:01.17 | blitzrage | Druken: vm-sorry? |
| 16:01.27 | lanec | is Mini-Itx a very poor form factor for an asterisk box because of its lack of pci slots? |
| 16:01.37 | Druken | blitzrage: oh yeah.. hehe i forgot about that one |
| 16:01.49 | blitzrage | actually... I have a im-sorry prompt and this is just stock asterisk with extra sounds file |
| 16:02.03 | blitzrage | ls /var/lib/asterisk/sounds/*sorry* |
| 16:02.27 | blitzrage | lanec: depends if you need to install a bunch of cards |
| 16:02.35 | blitzrage | if that is the case... then I would say yes |
| 16:02.36 | Druken | pfft... guess i should have checked the damn files i have before asking eh? |
| 16:02.43 | blitzrage | Druken: possibly :) |
| 16:03.49 | lanec | Sorry my question would yield a more accurate answer if I had stated it correctly. |
| 16:04.02 | lanec | Are there USB cards or just PCI cards for an asterisk box |
| 16:04.30 | benjk | there is a USB channel bridge, its called Astribank |
| 16:04.40 | benjk | channel bank |
| 16:04.55 | blitzrage | zoa had another one that he liked more at AstriCon, but I can't remember the name of it |
| 16:06.01 | benjk | also there is a USB based single-port BRI interface |
| 16:08.47 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 16:10.21 | _BOBWEEVER | If the codec stays the same, does SIP => IAX conversion use a lot of cpu? |
| 16:11.33 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 16:14.01 | Druken | blitzrage: know where i might find 'networks' ?? :) |
| 16:14.52 | *** join/#asterisk lorinc (n=ang@caracas-3927.adsl.interware.hu) |
| 16:19.43 | blitzrage | Druken: record it? |
| 16:20.05 | blitzrage | if you have all the sound prompts installed, and its not listed... seems pretty obvious it might not exist |
| 16:22.37 | *** join/#asterisk dacleric (n=dacleric@p5482237A.dip0.t-ipconnect.de) |
| 16:23.39 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-168-170.dhcp.southerncoastalcable.net) |
| 16:24.14 | *** join/#asterisk humbug (n=user@000-033-553.area3.spcsdns.net) |
| 16:25.13 | QbY | I'm encountering a problem, hopefully someone has some experience with. I need to conect a caller to an outside number, as soon as the other number answers (its an IVR) I need it to send a DTMF 0. I have this working fine with the D option, however the toll free number I connect them to for some reason doesn't actually send an "ACK" which tells Asterisk the call has been answered, yet the call has. Any suggestions? |
| 16:27.27 | danp | i'm not familiar with the ACK part...how does that work wit other calls? |
| 16:27.37 | blitzrage | QbY: what do you want Asterisk to do about the other end not sending an ACK? |
| 16:28.29 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 16:28.44 | QbY | blitzrage. I'd like it to go ahead and send the dtmf 0. What's funny is, the call is set up and it is passing audio. However, in the CLI you don't see SIP/blah answered ... you only see SIP/blah is making progress.. |
| 16:28.54 | QbY | I found its not sending an ACK by dumping the traffic.. |
| 16:28.56 | danp | ahh |
| 16:28.57 | *** join/#asterisk hi365_ (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 16:29.23 | file | what's the tollfree number for? |
| 16:29.25 | blitzrage | QbY: right... so Asterisk isn't going to progress unless the other call answers the call. You'd have to hack the code to progress where it doesn't expect to |
| 16:29.31 | QbY | file: for ADT. |
| 16:29.32 | blitzrage | at least that'd be my guess |
| 16:29.52 | file | they are cheating so they don't have to pay probably and sending some audio as inband progress until the last possible moment they can |
| 16:30.12 | QbY | file. that's what i'm guessing. because when i call any thing else, it works fine. |
| 16:30.59 | danp | i'm just trying to understand...are you calling them via SIP or through a SIP provider? |
| 16:31.26 | QbY | danp. through a SIP provider, I have also tried IAX2 but it is exhibiting the exact same behavior. |
| 16:32.10 | blitzrage | file: !!! |
| 16:32.26 | danp | huh. sounds funky |
| 16:36.44 | *** join/#asterisk joelsolanki (i=joelsola@220.224.90.134) |
| 16:38.15 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 16:39.10 | humbug | I would like to be able to execute some additional commands after a voicemail is left, but none of the exten's below Voicemail() get executed. |
| 16:39.38 | blitzrage | humbug: use the 'h' extension |
| 16:40.36 | EmleyMoor | If my standard ring cadence is 0,0,0, should I be worried? |
| 16:41.20 | EmleyMoor | Also, how do I dial # with SJphone? (as part of the number) |
| 16:41.30 | EmleyMoor | asterisk is changing it |
| 16:41.35 | humbug | blitzrage: the only options I see on the wiki are 's' 'u', and 'b' |
| 16:41.53 | blitzrage | humbug: the wiki has errors sometimes |
| 16:42.07 | blitzrage | humbug: 'h' is for processing stuff after a call ('h'angup) |
| 16:42.14 | humbug | ahh... perfect |
| 16:42.15 | humbug | thanks |
| 16:42.45 | EmleyMoor | Anyone around on BT? |
| 16:45.32 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 16:47.09 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 16:50.48 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 16:50.50 | Stephnie | hi |
| 16:50.58 | Stephnie | Dec 17 07:46:35 WARNING[18054]: acl.c:244 ast_get_ip_or_srv: Unable to lookup 'sip.broadvoice.com' |
| 16:51.14 | Stephnie | I get this msg when I issue "sip reload" |
| 16:52.01 | cpm | dns issue? |
| 16:52.08 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 16:52.09 | Stephnie | but when I ping sip.broadvoice.com ...I get ping reply.... |
| 16:52.09 | QbY | can you ping hostnames? |
| 16:52.17 | Stephnie | yes. |
| 16:52.17 | QbY | interesting. |
| 16:52.57 | Rawplayer | srvlookup = yes |
| 16:53.03 | Rawplayer | do you have that? |
| 16:53.09 | Stephnie | yes |
| 16:53.13 | cpm | on your asterisk server, what is the hosts: line in your nsswitch.conf file? |
| 16:53.41 | Stephnie | in /etc/hosts ? |
| 16:53.54 | cpm | no, nsswitch.conf |
| 16:54.02 | cpm | there is a line 'hosts:' |
| 16:54.10 | Stephnie | ok |
| 16:54.12 | Rawplayer | hosts: files dns |
| 16:54.17 | Rawplayer | that should be there |
| 16:54.28 | Stephnie | hosts: files dns |
| 16:54.40 | cpm | just for fun, reverse the order to dns files |
| 16:54.47 | Stephnie | everything was fine till last nite.. |
| 16:54.51 | cpm | hrmm |
| 16:55.09 | Rawplayer | what happens when you create a mapping in /etc/hosts? |
| 16:55.14 | Rawplayer | for that host |
| 16:55.32 | Stephnie | I tried to do that too...but same problem... |
| 16:55.45 | Stephnie | broadvoice told me to put entry in hosts. |
| 16:55.59 | QbY | What would cause * to ignore the # coming from the called party when the Dial command has the t? |
| 16:57.39 | Stephnie | if this problem doesnt get resolved..then I must changed my dedicated server provider. |
| 16:57.54 | Stephnie | change* |
| 16:59.10 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 16:59.16 | QbY | I'd change my SIP provider first. I hate Broadvoice. |
| 16:59.17 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 17:00.21 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 17:00.21 | Stephnie | QbY: I hate Broadvoice too...so whats your SIP Provider now? |
| 17:00.39 | QbY | Stephnie. We use a bunch. Including ourselves now. |
| 17:01.05 | *** join/#asterisk ozoneco (n=stanp@CPE-24-27-138-124.neb.res.rr.com) |
| 17:01.29 | Stephnie | I wish I could get a low rate routes for US and Canada ...I better kick this broadvoice out then. |
| 17:01.57 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 17:02.16 | QbY | Some that I'm happy with, and didn't cost a fortune to get in or high monthly rates, are http://connect.voicepulse.com, http://www.vitelity.net, and http://www.gafachi.com |
| 17:02.32 | EmleyMoor | I need some help from soneone in the UK with a BT line |
| 17:02.38 | EmleyMoor | someone,even |
| 17:03.50 | *** join/#asterisk Winkie (n=urmom@host86-130-188-100.range86-130.btcentralplus.com) |
| 17:04.43 | *** join/#asterisk apardo (n=apardo@87.217.144.227) |
| 17:05.35 | Rawplayer | Stephnie: what if you use proxy.chi.broadvoice.com |
| 17:05.50 | Rawplayer | instead of sip.broadvoice.com |
| 17:06.03 | Rawplayer | sip.broadvoice.com is a CNAME for proxy.chi.broadvoice.com |
| 17:07.00 | *** join/#asterisk Winkie (n=urmom@host86-130-188-100.range86-130.btcentralplus.com) |
| 17:12.59 | Stephnie | Rawplayer: if I use proxy.chi.broadvoice.com or IP Address then lines doesnt get registered. |
| 17:13.16 | Stephnie | Rawplayer: SIP.broadvoice.com is a must .. |
| 17:13.43 | Stephnie | I have tried it in X-lite softphone too |
| 17:13.55 | Stephnie | it only works with sip.broadvoice.com |
| 17:13.56 | QbY | then it is probably broadvoice |
| 17:14.04 | QbY | call them, and wait an hour on hold.. |
| 17:14.08 | QbY | so they can tell you nothing. |
| 17:14.09 | QbY | :) |
| 17:14.22 | QbY | i hate them. |
| 17:14.28 | Rawplayer | what if you create a zone file on your local dns server |
| 17:14.32 | Stephnie | yeah I called them...he said ..he never tried to use ip ..then he used ip |
| 17:14.32 | Rawplayer | for broadvoice.com |
| 17:14.36 | Stephnie | but ...failed |
| 17:14.52 | Rawplayer | and then create sip.broadvoice.com with the ip |
| 17:15.10 | Stephnie | so...then he came up with ....sip.broadvoice.com must be used... |
| 17:15.36 | Stephnie | Rawplayer : create sip.broadvoice.com ? |
| 17:15.45 | Rawplayer | yes.. |
| 17:15.46 | Stephnie | sorry..could get you |
| 17:15.51 | Rawplayer | on your local dns server |
| 17:15.52 | Stephnie | couldnt |
| 17:15.55 | Rawplayer | k |
| 17:16.59 | Stephnie | create sip.broadvoice.com for ? |
| 17:21.30 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 17:21.42 | EmleyMoor | I need some practical help from someone on BT |
| 17:23.44 | *** join/#asterisk isladelobos (n=user@43.Red-83-38-108.dynamicIP.rima-tde.net) |
| 17:24.55 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 17:24.58 | *** join/#asterisk toxap (i=toxap@194.187.128.88) |
| 17:26.03 | *** join/#asterisk toxap (i=toxap@194.187.128.88) |
| 17:27.02 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 17:27.14 | toxap | Hello |
| 17:29.48 | QbY | hrmmm.. What would cause Asterisk to detect DTMF over an IAX2 channel and not SIP |
| 17:29.52 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 17:30.11 | danp | is your dtmfmode set correctly |
| 17:30.18 | QbY | i've tried all |
| 17:30.24 | QbY | auto, inband, rfc... |
| 17:31.59 | toxap | When I compiling zaptel (zaptel-1.4.0-beta3), I have a problem. I do make and have error: |
| 17:31.59 | toxap | configure: *** Zaptel build successfully configured *** |
| 17:31.59 | toxap | **** |
| 17:31.59 | toxap | **** The configure script was just executed, so 'make' needs to be |
| 17:32.00 | toxap | **** restarted. |
| 17:32.02 | toxap | **** |
| 17:32.04 | toxap | make: *** [config.status] Error 1 |
| 17:32.06 | toxap | What this? |
| 17:33.03 | xheliox | benjk: This card I was talking about has that Maxim-IC as you said... does that give you any clues as to who made it? |
| 17:33.20 | toxap | sorry, I badly know english |
| 17:35.48 | toxap | Help me please, with zaptel |
| 17:39.09 | *** join/#asterisk jbrock (n=root@cpe-75-179-164-7.woh.res.rr.com) |
| 17:39.47 | isladelobos | l |
| 17:39.50 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 17:40.20 | jbrock | good afternoon all |
| 17:41.21 | isladelobos | good |
| 17:42.42 | *** join/#asterisk JVH (n=jvh@CPE-24-163-223-125.kc.res.rr.com) |
| 17:44.08 | oQpA | h |
| 17:45.26 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:45.30 | PakiPenguin | jmmm |
| 17:46.36 | oQpA | g |
| 17:46.42 | jbrock | what new features were added to 1.2.14 ? |
| 17:47.06 | oQPa | gf |
| 17:47.23 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 17:49.02 | oQPa | lol |
| 17:49.56 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:51.30 | EmleyMoor | What's the easiest way to tell if a call is internal or from outside in the dialplan? |
| 17:52.28 | *** join/#asterisk jbrock (n=root@cpe-75-179-164-7.woh.res.rr.com) |
| 17:52.36 | JVH | Hi all, trying to setup an Asterisk Testbed for our new phone system. Not being a Linux person the learning curve is somewhat steep. |
| 17:52.38 | EmleyMoor | (Outside calls come in over Zap/4, SIP and IAX2) |
| 17:54.52 | JVH | I have 2 phones Aasta 9133i and a Grandstream GXP2000. For the inital setup I am using Trixbox with Centos 4. The Aastra loads fine but the Grandstream says NO IP. The tech from Grandstream says the Bootp server is not running. I don't find a bootp server on the system. Is this a function of Dhcpd? |
| 17:57.21 | tzafrir | toxap, just like the text says. This is not really an error. Just a way to make you re-run make. |
| 17:57.49 | tzafrir | Should be a workaround some buggy make implementations that require re-running make separately |
| 17:58.18 | tzafrir | JVH, no |
| 17:58.25 | tzafrir | JVH, what distro is it? |
| 17:58.39 | tzafrir | JVH, netstat -lnup | grep 69 |
| 17:58.48 | toxap | tzafrir, When I re-run make, this problem repeats oneself |
| 17:59.02 | EmleyMoor | If a Goto (or GotoIf) happens, into a new context, do the variables for the call remain available? |
| 17:59.16 | tzafrir | toxap, what do you get when you run 'make -n' ? |
| 17:59.40 | tzafrir | toxap, did you see any warnings about a skewed clock? |
| 18:00.11 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:00.16 | toxap | tzafrir, Yes, make: Warning: File `Makefile' has modification time 2.8e+07 s in the future |
| 18:00.34 | tzafrir | toxap, what is the output of 'date' ? |
| 18:00.48 | toxap | tzafrir, Tue Jan 17 22:01:04 MSK 2006 |
| 18:00.58 | *** join/#asterisk humbug (n=user@000-025-078.area3.spcsdns.net) |
| 18:01.00 | JVH | Trixbox 1.2.3 Centos 4 |
| 18:01.37 | EmleyMoor | What are local variables local to? |
| 18:01.43 | tzafrir | toxap, get your clock straight. e.g: ntpdate -u pool.ntp.org |
| 18:02.09 | tzafrir | toxap, you're almost a yeaqr behind the rest of us. |
| 18:02.28 | EmleyMoor | Living in the past |
| 18:02.41 | naftali5 | and hurry up or you'll miss the holidays! |
| 18:03.43 | JVH | tzafrir I entered that command I am not sure what the results mean |
| 18:03.49 | tzafrir | JVH, if that netstat command gave any output, you have a tftp server (tftp uses UDP port 69) |
| 18:04.15 | tzafrir | JVH, hmm, my mistake |
| 18:04.35 | tzafrir | JVH, it probably also picked up IAX2 (4569) |
| 18:04.49 | tzafrir | netstat -lnup | grep -w 69 |
| 18:04.53 | humbug | I would like to execute some additional commands after a voicemail is successfully left, but nothing in the macro after the voicemail() gets run. Someone mentioned the 'h' option, but I can't find anything about it (not even in the source - I'm using 1.2). |
| 18:05.01 | toxap | tzafrir, Thanks, there was a problem in time |
| 18:05.33 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:05.36 | *** join/#asterisk Alter-Ego (n=chatzill@p1n22.ruraltel.net) |
| 18:05.44 | tzafrir | humbug, the user hangs up? |
| 18:05.57 | humbug | yeah |
| 18:06.04 | tzafrir | the 'h' extension gets executed when the call hangs up |
| 18:06.09 | *** part/#asterisk Alter-Ego (n=chatzill@p1n22.ruraltel.net) |
| 18:06.25 | humbug | ahhh... I was thinking it was a voicemail option |
| 18:06.42 | *** join/#asterisk Fibersrv (n=IceChat7@66-189-233-17.dhcp.cpgr.mo.charter.com) |
| 18:06.52 | JVH | I get udp 0.0.0.0 2424/xinetd |
| 18:07.22 | humbug | tzafrir: so it would be something like exten => h,n, command()? |
| 18:07.37 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:07.41 | tzafrir | JVH, that one is irrelevant |
| 18:08.18 | *** part/#asterisk Fibersrv (n=IceChat7@66-189-233-17.dhcp.cpgr.mo.charter.com) |
| 18:11.39 | tzafrir | JVH, ask some centos people here. I'm not sure if the tftpd runs independently or with xinetd |
| 18:12.23 | tzafrir | JVH, hmmm... my mistake again. It seems that xinetd is indeed listening on port 69. Anyway, I got to go now |
| 18:12.52 | JVH | ok thanks |
| 18:14.19 | *** part/#asterisk humbug (n=user@000-025-078.area3.spcsdns.net) |
| 18:14.46 | xheliox | Is there anyone in Dallas with a true blue Digium or Sangoma T1 (doesn't matter how many ports, one or more) card in stock that can be picked up today? |
| 18:15.10 | xheliox | Or first thing in the morning. |
| 18:15.13 | *** join/#asterisk double0 (n=chatzill@88-149-169-165.f5.ngi.it) |
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| 18:32.00 | *** join/#asterisk resistance (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 18:32.10 | resistance | hello everyone |
| 18:32.24 | resistance | i've been doing some reading about deadagi |
| 18:32.29 | *** join/#asterisk nvrs (n=RUR@bas5-kitchener06-1096638771.dsl.bell.ca) |
| 18:32.41 | resistance | i need something that will execute after hangup |
| 18:33.05 | resistance | will i be able to initiate a dial command from the dadagi script |
| 18:33.13 | resistance | ~dead |
| 18:33.18 | PakiPenguin | dadagi? |
| 18:33.19 | PakiPenguin | lol |
| 18:33.47 | resistance | whose u're daddy? |
| 18:33.53 | resistance | lol |
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| 18:41.52 | *** part/#asterisk JVH (n=jvh@CPE-24-163-223-125.kc.res.rr.com) |
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| 18:46.05 | resistance | if i pass an argument into php agi: AGI(agi.php|23), how so I access it in the script? |
| 18:48.41 | *** join/#asterisk obnauticus (i=asd@c-24-21-91-140.hsd1.wa.comcast.net) |
| 18:52.15 | resistance | if i pass an argument into php agi: AGI(agi.php|23), how so I access it in the script? |
| 18:52.25 | blitzrage | resistance: yes, you could execute another Dial() from the AGI script |
| 18:52.38 | blitzrage | resistance: repeating is unnecessary |
| 18:53.03 | blitzrage | $incoming_arg = $argv[1]; |
| 18:53.24 | blitzrage | ~book |
| 18:53.26 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:53.30 | blitzrage | read the AGI chapter |
| 18:53.33 | resistance | blizrage: i thought someone else might have come on that could help me out |
| 18:53.35 | resistance | thanks |
| 18:58.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-81-78.red.bezeqint.net) |
| 19:00.41 | _BOBWEEVER | Can anyone here tell me if * can connect to an mysql server on another box for realtime? |
| 19:00.58 | _BOBWEEVER | I thought I read somewhere that you could not, but it appears that you can in res_mysql.conf |
| 19:02.51 | blitzrage | I know in ODBC it doesn't matter where the DB server is |
| 19:03.04 | blitzrage | so if you're using res_odbc, no problem doing that |
| 19:05.29 | _BOBWEEVER | Thanks.. I will probably go that route. I have never had a problem with res_mysql on the same box, however I am unable to connect to a mysql instance on another box. |
| 19:06.10 | Nugget | I'm pleased to have gotten res_pgsql going last night. Looking forward to 1.4 release. |
| 19:06.33 | Nugget | I'm running the beta on my home system for now, but not on my production boxes. |
| 19:07.40 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 19:10.52 | _BOBWEEVER | I dont even see mysql traffic in the tcpdump on the target server.. |
| 19:12.47 | _BOBWEEVER | :q |
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| 19:14.18 | *** join/#asterisk mcquaid (n=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 19:14.50 | nvicf | hello there |
| 19:15.32 | mcquaid | hello, not asterisk related, but i'm about to unlock an ata and wondered if anyone had experience with that |
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| 19:57.46 | *** join/#asterisk NeScHe`mUziK`diN (n=_TugChe_@85.107.171.252) |
| 19:58.34 | SheriF_SpacE | why no one updated asterisk.org news with the new asterisk releases ? |
| 19:58.50 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 20:00.26 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
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| 20:02.18 | Nugget | because file is a slacker. |
| 20:03.25 | Un1x | im a slacker |
| 20:03.28 | Un1x | its why i use slackware |
| 20:03.35 | Nugget | I used to be but I sold the domain. :( |
| 20:03.42 | SheriF_SpacE | i see the post are astersikteam where is the rest of the team |
| 20:04.01 | SheriF_SpacE | Un1x: no u use slackware cuz u don't need a pacakge manager :P |
| 20:04.20 | Un1x | heh that too |
| 20:04.27 | Un1x | i prefer compiling from source i dont mind it |
| 20:04.39 | Nugget | slackware is the least linuxy linux. that's why I love it. |
| 20:05.03 | SheriF_SpacE | Un1x: me too but i don't have a powerful machine or fat connection , if i do i'll use Gentoo |
| 20:05.07 | SheriF_SpacE | ;-) |
| 20:06.24 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
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| 20:47.27 | *** join/#asterisk savage1 (n=savageon@66.212.207.34) |
| 20:47.30 | savage1 | hello all |
| 20:50.05 | Nugget | moo |
| 20:51.28 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:54.15 | nvicf | I have no idea how to setup this voip in my asterisk, I only have user/pass/ip, nothing more, I have some interns, and I don't have a clue, there's a lot of voip resources for this, it's a mess |
| 20:55.55 | *** join/#asterisk Omer (i=Omer@203.81.233.202) |
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| 20:58.01 | *** mode/#asterisk [+o denon] by ChanServ |
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| 21:11.18 | hmmhesays | well the vikings tanked again |
| 21:11.28 | *** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net) |
| 21:11.55 | nvicf | heh |
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| 21:31.07 | dknight11 | Is there any information available on the 1.4 Shared Line Appearance (SLA) feature beyond sla.conf.sample and comments in app_meetme.c? |
| 21:32.05 | *** join/#asterisk foxxtrot_ (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 21:32.40 | masonc | GUI question: I have a plain vanilla asterisk install going in today to a lawyers office. The systems admin at the office is not capable of root access, but I would like to give him the ability to change the extensions info, especially email addresses for voicemail. The system is configured, I don't want to startt again with a freepbx install. What GUI would work for this? |
| 21:34.52 | hmmhesays | you could try the 1.4 gui |
| 21:35.34 | Druken | masonc: i reccomend a custom built one :) |
| 21:40.17 | Strom_C | masonc: it's a lawyer's office for god's sake. sell them a maintenance plan :) |
| 21:40.23 | enema_cow | clear |
| 21:40.32 | hads | That'sa much better idea. |
| 21:40.44 | enema_cow | wrong terminal... :P |
| 21:40.48 | hads | Lawyers love that sort of thing |
| 21:40.49 | enema_cow | hi all |
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| 21:45.19 | *** join/#asterisk apardo (n=apardo@87.217.144.227) |
| 21:48.12 | orlok | hmmm |
| 21:49.28 | *** join/#asterisk betatester (n=tester@pool-71-251-229-244.rcmdva.fios.verizon.net) |
| 21:49.57 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 21:52.21 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 21:53.40 | lenne_dk | Access to voicemail prompts: Is there a more portable way to access the greeting from the mailbox than /var/spool/asterisk/voicemail/default/${number}/greet ? |
| 21:54.39 | tzafrir_laptop | OT: here's the latest spam I got. Nice try: http://pastebin.ca/282816 |
| 21:56.22 | orlok | Does anybody here know about/how to use SIP_HEADER? |
| 21:56.37 | *** join/#asterisk bluregard (n=bluregar@c-67-163-72-68.hsd1.il.comcast.net) |
| 21:56.39 | orlok | tzafrir_laptop: bwahaha! thats a way to defeat OCR |
| 21:56.54 | *** part/#asterisk bluregard (n=bluregar@c-67-163-72-68.hsd1.il.comcast.net) |
| 21:59.19 | resistance | i can't seem to get this: how can i do a max ringtime with AGI: $AGI->exec("DIAL zap/channel,30") |
| 22:03.56 | *** join/#asterisk [Airwolf] (n=airwolf@84.241.223.253) |
| 22:05.34 | *** join/#asterisk CleanerX (n=nix@p54A38666.dip0.t-ipconnect.de) |
| 22:06.06 | masonc | There will be a maintenance plan - but the guy likes to tinker |
| 22:08.23 | dknight11 | so...no information available on Shared Line Appearances? |
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| 22:09.54 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
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| 22:18.35 | lenne_dk | What do you want to know about SIP_HEADER? |
| 22:19.05 | orlok | As far as i can see in the docs,it only allows for getting sip header information |
| 22:19.21 | orlok | i am wondering if you can set it as well, so i can set the Invite address to be the same as the To: address |
| 22:19.25 | lenne_dk | exten => s,1,Set(SIP_HEADER(headername)=Foo Fighters) |
| 22:19.59 | lenne_dk | See http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header |
| 22:20.13 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 22:20.14 | file | that wiki page is wrong |
| 22:20.23 | file | you can't write to the SIP_HEADER dialplan function |
| 22:20.29 | lenne_dk | Then fix it :-) |
| 22:20.46 | lenne_dk | Either the wiki or asterisk |
| 22:20.56 | file | the Asterisk documentation has already been fixed |
| 22:20.58 | orlok | well, damn, cos asterisk is using the wrong data to determine the number the inbound call is for |
| 22:21.52 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 22:22.10 | lenne_dk | strange it only happens to you, orlok... Sure you are not doing anything wrong? |
| 22:22.25 | file | orlok, are you registering somewhere and then they are sending calls to you? |
| 22:23.54 | *** join/#asterisk JunK-Y (n=junky@modemcable198.14-83-70.mc.videotron.ca) |
| 22:24.19 | orlok | lenne_dk: n, it does seem to happen to other people |
| 22:24.25 | orlok | file: yup |
| 22:24.48 | file | then they are using the Contact URI as the Request URI in the INVITE |
| 22:25.07 | orlok | file: one sip account, 10 DID's, inbound calls all have the same INVITE: DID, the To: number is the one the call is actually inbound on |
| 22:26.41 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 22:27.04 | lenne_dk | Isn't the Wiki page example part of what you need? |
| 22:27.08 | lenne_dk | exten => +49123456789,1,Set(DN=${SIP_HEADER(TO):5}) |
| 22:27.08 | lenne_dk | exten => +49123456789,2,Set(DN=${CUT(DN,@,1)}) |
| 22:27.23 | lenne_dk | then jump on the DN |
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| 22:31.02 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
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| 22:34.49 | _BOBWEEVER | Has onyone here noticed choppy prompts with 2.6.9-42 kernel and vmware? |
| 22:35.52 | lenne_dk | orlok, can you use this idea? |
| 22:37.01 | *** join/#asterisk tinrsh (n=claudiu@81.181.94.112) |
| 22:37.16 | tinrsh | 'nite |
| 22:37.50 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 22:38.35 | Dovid | morning all |
| 22:39.00 | rudholm | good afternoon |
| 22:39.03 | xheliox | good evening |
| 22:39.07 | Dovid | ;) |
| 22:39.20 | tinrsh | hi there, any one worked here with audiocodes mp-118 fxo ? |
| 22:39.26 | Dovid | :) |
| 22:39.33 | Dovid | :( - sorry |
| 22:39.37 | Nugget | happy festivus |
| 22:39.52 | Dovid | happy hanulah and soon to be merry christmas |
| 22:39.57 | _BOBWEEVER | ahh.. a festivus for the restofus |
| 22:40.36 | xheliox | Bah humbug. |
| 22:41.49 | tinrsh | could someone point me on reading or something about how to choose a specific port for dial-out, but still be able to select a trunk with a random free line ? |
| 22:41.57 | tinrsh | on AC MP-118 FXO ? |
| 22:42.05 | Dovid | specific yet random ? |
| 22:42.10 | Dovid | is than an oxymoron ? |
| 22:42.43 | tinrsh | no |
| 22:42.46 | Dovid | isnt tha( |
| 22:42.53 | Dovid | amaybe i am jsut real tired |
| 22:43.21 | tinrsh | dependind on the extension that is dialing I have to choose a specific port for dial out, or an random one |
| 22:43.39 | tinrsh | like some users have to "go out" on a specific line |
| 22:43.52 | Dovid | ok. |
| 22:44.19 | Dovid | then u can group the users in diffrent context's based on what u want |
| 22:44.28 | tinrsh | yes |
| 22:44.35 | Dovid | so users 10-20 will go out thru random |
| 22:44.42 | Dovid | and 20-30 will have specific ports |
| 22:44.49 | Dovid | gorup them by diffrent context's |
| 22:45.19 | tinrsh | yes, but the random ports should also include the ones that are used by the 10-20 group |
| 22:45.26 | Dovid | ok |
| 22:45.41 | Dovid | so u use the canisavail command to see if the channel is available |
| 22:45.43 | Dovid | chanisavail* |
| 22:46.21 | tinrsh | my problem si the tunks and endpoints on mp-118, I really don't get it how to do that |
| 22:46.47 | tinrsh | s/si/is/ |
| 22:47.09 | *** join/#asterisk evilbuny (n=evilbuny@202.10.81.200) |
| 22:47.28 | Dovid | mp-118 is ? |
| 22:48.02 | tinrsh | AudioCodes MP-118 FXO |
| 22:48.13 | Dovid | too tired to really think |
| 22:48.17 | Dovid | been up for 3 days |
| 22:48.25 | tinrsh | ok, thanks for trying |
| 22:48.25 | Dovid | is it a gateway or card |
| 22:48.31 | Dovid | ? |
| 22:48.33 | tinrsh | a gateway |
| 22:48.44 | Dovid | ah |
| 22:48.49 | Dovid | sip to pots ? |
| 22:48.54 | tinrsh | yes |
| 22:49.03 | Dovid | ok |
| 22:49.10 | Dovid | so y not try this |
| 22:49.19 | Dovid | set a diffrent sip acocunt for every pots line |
| 22:49.29 | Dovid | and then when a call is made set a vairable to say 1 |
| 22:49.34 | Dovid | so the logic is |
| 22:49.54 | Dovid | if SIP1 is in use (meaning pots1 is in use) then set flag to in use |
| 22:50.07 | Dovid | now when u want to make a call check first to see if the flag is set to in use |
| 22:50.14 | tinrsh | oh... yeah good ideea |
| 22:50.18 | Dovid | if it is then go to next sip account |
| 22:50.31 | Dovid | if it isnt than set flag to in use and use that sip account |
| 22:50.48 | Dovid | when when the call ends use the h extension to reset the variable |
| 22:50.52 | tinrsh | I did one account per gateway, and I was thinking to manipulate it from the trunking and routing in the gateway |
| 22:51.09 | tinrsh | but you ideea seems more reliable at this moment |
| 22:51.11 | tinrsh | thanks |
| 22:51.19 | Dovid | i perosonally think its better when you have more control on the asterisk side |
| 22:51.21 | Dovid | yw |
| 22:51.23 | Dovid | its open source |
| 22:51.27 | Dovid | we help one another ;) |
| 22:51.37 | Dovid | people have helped me and i feel its my duty to give back |
| 22:51.38 | Dovid | good luck |
| 22:52.21 | ucfMethod | anyone having issues with the Polycom 501s and the latest 2.0.3 sip.ld???? After I upgraded the phones, occasionally (almost every time i try) the buttons are not responsive, sometimes you have to press a number 2-4 times before it works. |
| 22:53.19 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:0:240:f4ff:fe6b:7650) |
| 22:56.53 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:59.16 | _BOBWEEVER | gain? |
| 23:03.22 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 23:03.58 | Ryanw | is there a way to listen to hold music through a speakerphone then have the music cut out when a call comes in? |
| 23:06.30 | IronHelix\AFK | its a function of the phone i think |
| 23:06.44 | IronHelix\AFK | a handful of phones have the option to connect to something when idle |
| 23:12.21 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:12.54 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:12.54 | EmleyMoor | My asterisk is detecting both of my available ring cadences as 0,0,0 |
| 23:13.03 | EmleyMoor | I'm on BT in the UK - why would that be? |
| 23:13.06 | IronHelix\AFK | do you have distinctive ring detection turned on? |
| 23:13.13 | EmleyMoor | Yes |
| 23:13.23 | IronHelix\AFK | hmmm |
| 23:13.26 | EmleyMoor | (in zapata.conf) |
| 23:13.31 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 23:13.36 | EmleyMoor | If it needs to be anywhere else as well, please advise |
| 23:14.22 | nowork | hello..I am in Toronto, any advise on where to buy a 1U server for *DID service? |
| 23:14.43 | EmleyMoor | (I am thinking of getting the third ring cadence but it will be of no use if asterisk refuses to tell the difference) |
| 23:14.47 | IronHelix\AFK | emley, what hardware? |
| 23:14.58 | EmleyMoor | Digium TDM400P |
| 23:15.26 | nowork | also think to use Asterisk to do phone card service,what kind of service should i know?what about Dell poweredge 1950? |
| 23:15.41 | IronHelix\AFK | and you have usedistinctiveringdetection=yes ? |
| 23:15.52 | EmleyMoor | IronHelix: Yes |
| 23:16.18 | IronHelix\AFK | and then you turn on verbose and it says 0,0,0? hmmm.... what signalling type? |
| 23:16.21 | EmleyMoor | It detects and displays the cadence as 0,0,0 in both the normal call and ringback service |
| 23:16.27 | EmleyMoor | fxs_ks |
| 23:16.44 | IronHelix\AFK | hmmm |
| 23:16.47 | IronHelix\AFK | that all sounds right |
| 23:17.13 | IronHelix\AFK | nowork- any decent box will do |
| 23:17.23 | IronHelix\AFK | it depends on what kind of line you get and if you are transcoding etc |
| 23:17.52 | IronHelix\AFK | em do you have all the localizations set to uk mode? |
| 23:18.15 | IronHelix\AFK | try pastebinning your zapata and zaptel.conf and i'll see if anything sticks out |
| 23:18.35 | EmleyMoor | IronHelix: As and where I know about them, yes - configs coming up soon |
| 23:23.01 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 23:30.24 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 23:31.41 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 23:35.00 | EmleyMoor | http://pastebin.ca/282927 |
| 23:40.26 | IronHelix\AFK | put loadzone and defaultzone above the channel definitions (fxoks=1 etc) |
| 23:40.44 | IronHelix\AFK | try turning off all the settings for distinctinve rings but leave usedistinctiveringdetection on |
| 23:40.46 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 23:41.07 | *** join/#asterisk BhaalWTF (n=bhaal@121.208.127.141) |
| 23:42.29 | IronHelix\AFK | ie turn off the custom ring cadences on lines 487-492, and put the dring context stuff after the normal context stuff (but comment it out for testing) |
| 23:44.04 | IronHelix\AFK | also try putting the dringcontext stuff below the signalling=fxs_ks part on line 581, perhaps its being cleared by the above *shrug* |
| 23:44.05 | IronHelix\AFK | bbl |
| 23:45.01 | EmleyMoor | Still detects 0,0,0 |
| 23:45.56 | *** join/#asterisk BhaalWK (n=bhaal@121.208.127.141) |
| 23:46.51 | EmleyMoor | The dringcontext stuff does work - but as it only detects 0,0,0 it's pretty pointless :-( |
| 23:51.42 | EmleyMoor | I need this to work soon :-( |
| 23:54.44 | *** join/#asterisk Farris1 (n=FarrisGo@pool-71-164-194-157.dllstx.fios.verizon.net) |
| 23:55.06 | Farris1 | Can anyone using SIP trunks from bandwidth.com help me figure out what I need to do to get dtmf working on inbound calls? |
| 23:59.03 | EmleyMoor | Anyone else here know about dring detection and BT? |