| 00:00.39 | Buttons840 | but what is a "call" i think asterisk has a different deffinition than most people; ie, i pick up the phone, call grandma, and talk for 5 minutes, then hang up the phone; this could be more than one call in the view of asterisk; thus multiple uniqueid's for what is commonly considered a single call |
| 00:00.47 | *** join/#asterisk QubeZ (~nkasu@68.204.67.110) |
| 00:00.50 | QubeZ | hello all |
| 00:01.23 | Buttons840 | hi |
| 00:01.29 | lost_soul | QubeZ: hey, hey |
| 00:02.03 | QubeZ | I have a gigabit blade in my cisco 6509 and port configured the same as my fastethernet, however when I move my Asterisk server (1.4.26.2) to gigabit, I cannot get dtmf tones to generate ie: cannot access vm because of this. When I move it back to fastethernet, all is working again. Using rfc standard for the dtmfmode. Any ideas? |
| 00:02.05 | Buttons840 | jdoe: i'm just making observations here; i'm not asking for help, though i'm interested in what others have to say |
| 00:06.00 | jdoe | Buttons840: not sure, but I'm also not sure under what circumstances you'd have a single channel having multiple cdrs other than explicitly forking one. |
| 00:07.21 | jdoe | or at least, I'm not sure what circumstances would give you that on a 1.6 version of asterisk. |
| 00:30.16 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 00:30.51 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 00:44.18 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 00:47.00 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 00:47.57 | ChannelZ | QubeZ: have you turned on debugging to see if it's getting dtmf and just ignoring it, etc? And what method of dtmf are you using? |
| 00:48.36 | ChannelZ | nevermind you said, rfc |
| 00:49.53 | ChannelZ | I can't think of a reason you would be getting bidirectional audio but no dtmf because it's data in the same stream. |
| 00:52.17 | QubeZ | ChannelZ: ya exactly, audio is perfectly ok but no dtmf |
| 00:52.45 | QubeZ | ChannelZ: i turned on dtmf on console and on gigabit it doesn't show up in the logs but on fastethernet i can see the dtmf tones |
| 00:53.01 | QubeZ | its really driving me nuts |
| 00:53.03 | pabelanger-lap | Any ideas? WARNING[13082]: config.c:2021 find_engine: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available |
| 00:54.39 | QubeZ | pabelanger-lap: http://www.hurdman.net/mirror/voip-info/wiki/view/Asterisk+RealTime.html |
| 00:54.44 | ChannelZ | you are trying to use realtime config but don't have your database engine fully configged? |
| 00:55.24 | pabelanger-lap | QubeZ: Thanks google ;) |
| 00:56.41 | pabelanger-lap | ChannelZ: Ya, not sure odbc show' |
| 00:56.54 | pabelanger-lap | ChannelZ: seems to be working |
| 01:08.43 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:35.13 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 01:36.20 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 01:39.08 | *** join/#asterisk Mhaddog_Mac (~anonymous@adsl-233-57-83.mia.bellsouth.net) |
| 01:43.00 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 01:52.32 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 01:55.03 | *** join/#asterisk Devon_ (~chatzilla@adsl-233-92-81.mia.bellsouth.net) |
| 01:56.09 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 01:57.54 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 01:59.31 | *** join/#asterisk geneticx (~geneticx@adsl-10-124-228.mia.bellsouth.net) |
| 02:05.32 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 02:15.05 | pabelanger-lap | damn you res_config_odbc.so |
| 02:18.39 | *** join/#asterisk coppice (~chatzilla@210.17.254.185) |
| 02:20.52 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 02:26.52 | *** join/#asterisk pls_wrk_fs (~Miranda@119.94.193.194) |
| 02:27.58 | pls_wrk_fs | hi. i have a call coming from the outside... it is recieved then for example by ext 100.. how can i bridge that call to ext 101 and passing the orig. CID of the caller? |
| 02:39.28 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 03:05.09 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 03:07.52 | *** join/#asterisk xpot-mobile (~james@c-24-2-70-81.hsd1.ut.comcast.net) |
| 03:14.01 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 03:18.17 | *** join/#asterisk QubeZ (~nkasu@110.67.204.68.cfl.res.rr.com) |
| 03:18.20 | QubeZ | hello all |
| 03:18.29 | QubeZ | is there a way for asterisk to monitor config files and reload if they are updated? |
| 03:20.47 | pwell | qube I think normally the OS might have something built in for that |
| 03:22.08 | QubeZ | pwell: i'll explore that, thanks |
| 03:22.12 | *** join/#asterisk path (~path@gateway/shell/bshellz.net/x-btrvvhgluxzqkwug) |
| 03:24.34 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 03:25.50 | WIMPy | inotify |
| 03:28.04 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 03:33.16 | pabelanger-lap | QubeZ: Asterisk realtime |
| 03:36.25 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 03:37.04 | ChannelZ | pls_wrk_fs: just Dial(SIP/101) (assuming that's what the device belonging to 'extension 101' is called) |
| 03:43.20 | QubeZ | pabelanger-lap: will look into it |
| 03:43.47 | QubeZ | how do i run command from the CLI for two consecutive ones in one line: asterisk -rx "logger reload" "dialplan reload" ? |
| 03:44.22 | p3nguin | Add another -rx before the second command. |
| 03:44.49 | QubeZ | asterisk -rx "logger reload" -rx "dialplan reload" <-- gotcha, thans |
| 03:51.31 | ChannelZ | rocks out with his clock out |
| 03:55.07 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 03:58.55 | xheliox | ChannelZ: Put your clock away, no one wants to see that. |
| 04:00.19 | ChannelZ | Where do I put it? It's very large. (bad eyesight) |
| 04:03.24 | *** join/#asterisk opus_ (~opus@72.52.102.105) |
| 04:17.42 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 04:25.48 | *** join/#asterisk mburlingame (~mike.burl@173-8-220-141-Oregon.hfc.comcastbusiness.net) |
| 04:49.23 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 04:52.00 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 04:56.17 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 05:07.48 | *** join/#asterisk Crymsonite (~lchrist20@ip68-104-111-21.lv.lv.cox.net) |
| 05:09.52 | Crymsonite | Don't know who's here and not. I have a couple of questions about AGI / Getting a PRI Line Routed to a SIP / IAX. |
| 05:10.04 | Crymsonite | If anyone could help out, it would be deeply appreciated. |
| 05:12.35 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 05:15.18 | [TK]D-Fender | Crymsonite: Ask a specific question, get a specific answer... |
| 05:22.39 | *** join/#asterisk VxJasonxV (~jason@xmms2/troll/VxJasonxV) |
| 05:23.13 | VxJasonxV | I'm trying to wrap my head around a callerid issue, and I'm not doing so well at it. We have a TDM2400p card plugged into our server, and no matter what I do, I can't seem to get the calling party's phone number. |
| 05:23.33 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 05:23.51 | VxJasonxV | My first question is: When someone external places a call, it comes in via DAHDI. Should I be seeing the phone number in the asterisk console with verbose/debug turned on and way up? |
| 05:24.09 | VxJasonxV | (someone external = a caller via POTS calls our main line) |
| 05:24.19 | VxJasonxV | and, verbose/debug = 10v/3d |
| 05:24.37 | [TK]D-Fender | VxJasonxV: No. NoOp it in your dialplan if you want to see it |
| 05:24.58 | [TK]D-Fender | (No, it will not SHOW in CLI by any built-in means. You have to do this) |
| 05:25.43 | p3nguin | Verbose() makes it show in the CLI, though. |
| 05:25.44 | [TK]D-Fender | VxJasonxV: pastebin your chan_dahdi.conf and everythign linked to it |
| 05:25.49 | [TK]D-Fender | ~pb |
| 05:25.50 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 05:32.56 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:34.04 | VxJasonxV | ok, that is a good starting point |
| 05:34.39 | VxJasonxV | NoOp(CALLERID(number)), right? |
| 05:34.49 | VxJasonxV | I'm not adding it in yet, but isn't that what it would be? |
| 05:35.30 | *** join/#asterisk Crymsonite (~lchrist20@ip68-104-111-21.lv.lv.cox.net) |
| 05:35.45 | Crymsonite | I have a AMI connection making an outbound call, gets connected. Now I want to ring a IAX client and upon answer, connect the two. I have done a sip bridge. But I have to have the SIP dialed into my dialplan and wait for the bridge. I perfer ring the IAX and bridge it that way. For some reason, I have tried a couple of things that made logical sense to me, but I cannot get it to work. |
| 05:36.03 | Crymsonite | Any tips on steering me in the right direction, would be appreciated. |
| 05:36.15 | p3nguin | Verbose(${CALLERID(num)}) if you want to print it on the CLI |
| 05:36.19 | p3nguin | vxjasonxv: ^^ |
| 05:36.48 | p3nguin | s/Verbose/NoOp/ if you want it only in the verbose messages. |
| 05:37.34 | VxJasonxV | ah, I was close :) |
| 05:37.37 | VxJasonxV | thanks |
| 05:37.53 | VxJasonxV | I think I might have just found my problem. |
| 05:38.04 | VxJasonxV | I have a callerid=asreceived , then a callerid= (blank) |
| 05:38.04 | [TK]D-Fender | [01:15]<[TK]D-Fender>Crymsonite: Ask a specific question, get a specific answer... |
| 05:38.14 | [TK]D-Fender | VxJasonxV: That would be a problem |
| 05:38.15 | p3nguin | You want to use the function as a variable, so you have to wrap it with ${} to parse the value. |
| 05:39.35 | VxJasonxV | http://asterisk.pastey.net/138236-18js |
| 05:39.48 | [TK]D-Fender | crimSounds like you're doing an Originate. So dial whichever you want first and then they get dumped into the dialplan wherever you tell it to and it will do whatever that dialplans ays, including dialing that other device |
| 05:39.51 | VxJasonxV | I'm sure you folks would agree. |
| 05:43.02 | *** join/#asterisk mrchrisadams (~Adium@CPE-121-217-155-70.lnse2.cht.bigpond.net.au) |
| 05:43.04 | *** join/#asterisk Crymsonite (~lchrist20@ip68-104-111-21.lv.lv.cox.net) |
| 05:43.57 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 05:45.02 | [TK]D-Fender | [01:39]<[TK]D-Fender>Crymsonite Sounds like you're doing an Originate. So dial whichever you want first and then they get dumped into the dialplan wherever you tell it to and it will do whatever that dialplans ays, including dialing that other device |
| 05:45.17 | [TK]D-Fender | Good grief... |
| 05:45.49 | VxJasonxV | he must use comcast |
| 05:46.00 | VxJasonxV | even though I can clearly see cox in the host |
| 05:46.05 | VxJasonxV | I still blame comcast |
| 05:48.17 | Slugs_ | hehe |
| 05:48.25 | Slugs_ | poor sack |
| 05:48.30 | VxJasonxV | Now I have to figure out how I wrangle FreePBX into submission. |
| 05:48.41 | Slugs_ | rm -rf |
| 05:48.46 | VxJasonxV | yeah, no kidding. |
| 05:48.49 | Slugs_ | ;) |
| 05:48.52 | VxJasonxV | Not my choice -_-. |
| 05:49.01 | Slugs_ | i hear that |
| 05:49.19 | VxJasonxV | I had a functional Asterisk installed on top of a nice clean CentOS instance, but "WE NEED A GUI" stepped in. |
| 05:49.23 | VxJasonxV | sighs |
| 05:49.28 | Slugs_ | being in it and working for non-technical managers is about as bad as being raped |
| 05:49.35 | VxJasonxV | well |
| 05:49.39 | path | why GUI if there is vi and emacs |
| 05:49.40 | VxJasonxV | I've never been raped, so I can't agree. |
| 05:49.49 | VxJasonxV | But, it would seem that would be slightly true. |
| 05:51.25 | VxJasonxV | yeah, doing his tonight is probably not a good idea... Because if I screw something up, that means that no one knows for 3 days |
| 05:51.29 | VxJasonxV | and that's... probably not good. |
| 05:51.48 | VxJasonxV | Thanks for the help [TK]D-Fender, and thanks for the laughs, Slugs_. Night everyone. |
| 05:51.52 | *** part/#asterisk VxJasonxV (~jason@xmms2/troll/VxJasonxV) |
| 06:16.03 | *** join/#asterisk Crymsonite (~lchrist20@ip68-104-111-21.lv.lv.cox.net) |
| 06:16.17 | Slugs_ | <[TK]D-Fender>Crymsonite Sounds like you're doing an Originate. So dial whichever you want first and then they get dumped into the dialplan wherever you tell it to and it will do whatever that dialplans ays, including dialing that other device |
| 06:16.46 | Crymsonite | Thank you, I dont know why I am having connection issues with IRC. appreciate the c&p |
| 06:16.53 | Slugs_ | np |
| 06:16.58 | Slugs_ | good night |
| 06:17.01 | Crymsonite | nite |
| 06:18.52 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 06:23.20 | bcrosby | what's the best way to not get asterisk to listen for incoming sip connections? |
| 06:26.38 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 06:27.24 | [TK]D-Fender | bcrosby: "noload => chan_sip.conf" <- modules.conf |
| 06:32.05 | bcrosby | ahh |
| 06:32.13 | bcrosby | but then I can't use SIP to my provider |
| 06:32.47 | ChannelZ | err... |
| 06:33.07 | ChannelZ | firewall it off then |
| 06:33.50 | [TK]D-Fender | bcrosby: then just specify a dead context in [general] |
| 06:49.42 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 06:52.41 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 06:53.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 07:12.31 | *** join/#asterisk redax (redax@r6.hu) |
| 07:12.32 | redax | hi |
| 07:13.52 | redax | is a stock asterisk 1.6.0.x + asterisk addons capable to log queue_log into mysql |
| 07:13.54 | ChannelZ | HAI! |
| 07:13.56 | redax | ? |
| 07:17.26 | ChannelZ | should be |
| 07:19.53 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 07:24.15 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:25.28 | redax | yep. it works. |
| 07:26.18 | redax | I can't find anything about the tablename configuration. is it hardcoded to `queue_log' ? |
| 07:29.28 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 07:30.28 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 07:33.33 | *** join/#asterisk Crymsonite (~Crymsonit@ip68-104-111-21.lv.lv.cox.net) |
| 07:38.40 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 07:40.01 | ChannelZ | redax: try |
| 07:40.13 | ChannelZ | argh x-chat's copy/paste is a piece of shit |
| 07:40.15 | ChannelZ | http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL |
| 07:56.49 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 07:59.02 | *** join/#asterisk markitoxs (~miranda83@cpc8-sgyl21-0-0-cust21.sgyl.cable.virginmedia.com) |
| 08:24.12 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 08:28.24 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 08:28.24 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 08:28.25 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 08:28.40 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 08:32.15 | *** join/#asterisk penguin_ice (~penguin@cpe-67-252-168-97.buffalo.res.rr.com) |
| 08:32.26 | penguin_ice | anyone around? |
| 08:32.40 | WIMPy | no |
| 08:33.05 | penguin_ice | damn its already 4:30 am here |
| 08:33.32 | ChannelZ | bollocks! |
| 08:38.23 | kami | Hello, I have upgraded a previous asterisk 1.4 installation on Ubuntu hardy to the latest one from the repository: 1.4.17~dfsg-2ubuntu1.1 |
| 08:39.11 | kami | And now the sound files are not found. |
| 08:39.46 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 08:39.47 | penguin_ice | did you check them in directory /var/lib/asterisk/sounds/en/ |
| 08:40.14 | penguin_ice | you can easily just download the sound files from source from the asterisk website and send them to that directory so asterisk can use them |
| 08:40.33 | Docteh | kami: check the dpkg log maybe dpkg removed a sounds package |
| 08:41.01 | kami | penguin_ice: by default, the installation still puts them in sounds/ and I didn't find a language_prefix=yes entry, so I thought asterisk should still be able to find them |
| 08:41.31 | kami | Docteh: the asterisk-sounds-main package is still there and there are .gsm files in /usr/share/asterisk/sounds |
| 08:42.06 | Docteh | I was wondering where that copy came from on my server |
| 08:42.33 | Docteh | does raising the debug level make it print out a full path? |
| 08:42.34 | kami | Docteh: you mean it doesn't belong there? |
| 08:43.22 | Docteh | well i used the package then switched to source, so i've got sounds in both /usr/share/asterisk and /var/lib/asterisk |
| 08:43.26 | kami | Docteh: I've tried with verbose 10 and debug 10 and it still only says 'No such file ...' I there any other debug config which I could change? |
| 08:44.01 | kami | Docteh: the ones in /var/lib/asterisk/sounds are symlinks to /usr/share ... on my system. Can you verify that? |
| 08:44.57 | Docteh | so if you ls in /usr/share/asterisk/sounds and /var/lib/asterisk/sounds theres something there? |
| 08:45.31 | kami | Docteh: yes, gsm files in /usr/share and non-dangling symlinks in /var/lib |
| 08:45.40 | kami | Docteh: but 'full path' is a good idea. I will configure the extension with a full path to the file and see what happens. |
| 08:45.45 | Docteh | hmm core set debug 1000 didn't work |
| 08:45.56 | Docteh | kami: leave off the .gsm iirc |
| 08:46.16 | p3nguin_ | What's the problem? |
| 08:46.19 | kami | Docteh: will do, thanks. |
| 08:47.46 | p3nguin_ | Actually, nevermind. |
| 08:47.57 | p3nguin_ | I just realized that I don't care. |
| 08:48.07 | p3nguin_ | Sorry for the inconvenience. |
| 08:48.24 | Docteh | :o |
| 08:50.43 | kami | That's strange. I get Unable to open /usr/share/asterisk/sounds/demo-echotest (format 0x2 (gsm)): No such file or directory |
| 08:54.10 | kami | Docteh: that was a nice one! My colleague who worked on the same problem must have moved only that single file demo-echotest.gsm to another dir (probably for testing). It was the only one which was NOT present in /usr/share |
| 08:54.33 | Docteh | lol |
| 08:54.35 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 08:54.55 | Docteh | you both picked the same file to play with, nice |
| 08:55.16 | ChannelZ | they call each other and dress the same too |
| 08:55.40 | kami | Docteh: he will no lol, any more when I meet him next time |
| 08:57.12 | kami | Well it was the 'Echo Test' extension which we tried to make run (to *simplify* a more complex scenario where the audio files from a VoiceXML app wouldn't play). Simplify ... hehe. |
| 08:57.35 | kami | Now, I can continue with the VoiceXML stuff. |
| 08:57.58 | kami | Docteh, penguin_ice: thank you! |
| 08:59.50 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 09:24.47 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:24.47 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 09:34.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 09:39.24 | *** join/#asterisk imcdona (~Administr@173.160.189.74) |
| 09:39.30 | *** part/#asterisk imcdona (~Administr@173.160.189.74) |
| 10:04.35 | *** join/#asterisk Markhoury (~tarekhour@194.165.158.108) |
| 10:05.46 | Markhoury | Hello mates, i need help configuring an SHDSL E1 |
| 10:05.50 | Markhoury | any1 can help? |
| 10:15.14 | Markhoury | ?? |
| 10:18.48 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:52.03 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 11:06.09 | *** join/#asterisk DexTerDDIT (~dexterddi@ppp2481078633.ambra.ro) |
| 11:20.01 | thevoke | b/win 49 |
| 12:16.44 | *** join/#asterisk jkroon (~jkroon@dsl-244-40-94.telkomadsl.co.za) |
| 12:26.16 | DexTerDDIT | ello i have question |
| 12:26.22 | DexTerDDIT | can some help ? |
| 12:30.17 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 12:31.16 | bn-7bc | DexTerDDIT: well ask the question and si if yo get an answer, I'll be glad to help if I can, but I don't know what the problem is? |
| 12:32.54 | DexTerDDIT | I heard i could you a motorola PCI Modem to connect the asterisk to my phoneline , can this be done ? |
| 12:33.09 | DexTerDDIT | *use |
| 12:34.10 | bn-7bc | hmm sorry I have just done sip so can anoneelse help |
| 12:35.43 | DexTerDDIT | I`m know the SIP part :) , o just want way to get the landline without digium hardware ... beacause it`s hard to get int these parts :P ... |
| 12:36.47 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 12:37.40 | jkroon | DexTerDDIT, i've hear rumours/seen hints that one can use normal ISDN modems (not analog), however, I haven't actually seen any concrete examples unless you can get CAPI/mISDN working on the card. |
| 12:39.46 | tzafrir_laptop | Markhoury, do you have it connected to a Linux system? |
| 12:41.01 | Markhoury | yes |
| 12:41.08 | Markhoury | CENTOS |
| 12:41.39 | bn-7bc | DexTerDDIT: try this http://www.voip-info.org/tiki-index.php?page=X100P+clone |
| 12:41.44 | DexTerDDIT | aha jkroom thanks |
| 12:41.57 | DexTerDDIT | ok thank you bn-7bc |
| 12:44.18 | jkroon | is somebody else able to open "https://issues.asterisk.org/file_download.php?file_id=26189&type=bug&download" ?? |
| 12:45.53 | bn-7bc | jkroon: well at least I can't "application error" |
| 12:46.40 | jkroon | bn-7bc, that page according to google is the only one on the interwebs containing "Changing from state 5 to 4" ... which is an error I'm running into at the moment. |
| 12:48.20 | tzafrir_laptop | Markhoury, how does that card show on 'lspci -v'? |
| 12:49.01 | Markhoury | let me check |
| 12:49.32 | jkroon | anybody else ever saw from pri show spans "PRI span 3/0: Provisioned, Down, Active" and what does it mean? |
| 12:49.37 | Markhoury | i have a sanogma pri card .. 4 spans |
| 12:49.46 | Markhoury | works fine with a normal E1 |
| 12:49.56 | Markhoury | but it doesn`t work with the SHDSL E1 |
| 12:50.05 | Markhoury | i get RED alarms |
| 12:50.32 | jkroon | and for that matter ... how can I fix it? |
| 12:53.29 | Markhoury | is there some setting that i need to change? |
| 12:54.31 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:54.38 | Markhoury | the card show when i do lspci yes |
| 12:56.08 | Markhoury | anyone have any idea how to configure an SHDSL E1? i have ASTERISK 1.6 and a Sanogma A104d card |
| 12:56.17 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:58.06 | tzafrir_laptop | Markhoury, oh, it's a Sangoma card? Which model, exactly? |
| 13:06.53 | Markhoury | A104d |
| 13:11.51 | Markhoury | tzafrir_laptop , the model is A104d |
| 13:12.08 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:12.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:14.53 | WIMPy | Markhoury: There shouldn't be any difference for you. Are you sure it's up? |
| 13:15.31 | Markhoury | i called the provider and told me that it`s up and running |
| 13:17.57 | Markhoury | could be cause the LOCATION has changed? |
| 13:23.20 | WIMPy | Let me read back... |
| 13:24.26 | WIMPy | Ok, no, if your setup works on another E1 it should wok the same on any other E1. |
| 13:25.13 | Markhoury | i thought so .. well this is weird .. i guess it`s a provider problem |
| 13:25.24 | WIMPy | The fact that it's delivered via shdsl only means that you will probably see a lot more sevice interruptions. |
| 13:27.22 | WIMPy | Does the modem give any indications? |
| 13:28.06 | Markhoury | nope .. there is only 1 blinking green light |
| 13:28.25 | Markhoury | i never got to work with an SHDSL before |
| 13:28.29 | WIMPy | Blinking? |
| 13:28.35 | Markhoury | ya |
| 13:29.27 | WIMPy | I usually wouldn't expect that to be a good sign, but unless that LED is labelled, that's hard to say. |
| 13:29.55 | Markhoury | it`s labled DCD/109 |
| 13:30.04 | Markhoury | which i have no idea what it is |
| 13:30.51 | WIMPy | Sounds like X35 or V29 to me. Any other LEDs? |
| 13:31.08 | Markhoury | nope |
| 13:32.12 | WIMPy | Not very helpfull. "Have you tried turning it off and on again?" |
| 13:32.25 | Markhoury | nope .. good idea |
| 13:32.28 | Markhoury | i will do that |
| 13:32.53 | WIMPy | Maybe that tells you more about what that LED says. |
| 13:35.19 | *** join/#asterisk coppice (~chatzilla@210.17.254.185) |
| 14:17.21 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 14:25.18 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 14:28.41 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:32.34 | Markhoury | exit |
| 14:32.51 | riddlebox | connecting to IP systems together, 1 Avaya IP office and the other Asterisk, would it be better to use a sip trunk or a sip extension on the IP Office? |
| 14:34.11 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 14:35.02 | Godfather_ | o/ |
| 14:45.54 | *** join/#asterisk dmz (~dmz@152.sub-97-18-172.myvzw.com) |
| 14:48.45 | *** join/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it) |
| 14:49.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:51.49 | *** join/#asterisk ndemir (~ndemir@94.121.187.94) |
| 14:53.26 | ndemir | hello. all traffic between two peers (ip client) goes through sip server (asterisk). the traffic should be directly between these two peers. am i wrong? |
| 14:55.53 | ndemir | is this issue about asterisk or sip clients? |
| 14:57.05 | [TK]D-Fender | ndemir: If * has a reason to stay in the path it will. If * is not told that each end is allowed to reinvite they won't |
| 14:58.28 | xheliox | [TK]D-Fender: Did you see the vote we took last night? :) |
| 14:58.55 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 15:00.20 | ndemir | [TK]D-Fender: so it seems about canreinvite parameter, because it is set to no in my configuration. |
| 15:00.38 | [TK]D-Fender | ndemir: great reason they won't do it then. |
| 15:00.55 | [TK]D-Fender | ndemir: Just remember that NAT on either end can screw that idea right up. |
| 15:01.50 | ndemir | <PROTECTED> |
| 15:03.48 | ndemir | <PROTECTED> |
| 15:04.07 | [TK]D-Fender | dnNo, I am not, and jsut ask. Whoever can & feels like answering will anyway |
| 15:11.26 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 15:14.37 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:22.44 | [TK]D-Fender | Or not. |
| 15:28.27 | riddlebox | hey [TK]D-Fender whats new? |
| 15:29.53 | [TK]D-Fender | riddlebox: Band is starting to get active. Show in 2 weeks, looking to book more. Single & hunting. and bucling down on the diet. |
| 15:30.17 | riddlebox | busy |
| 15:30.44 | riddlebox | I have been fighting connecting an IP Office to asterisk for a couple days down |
| 15:31.22 | riddlebox | [TK]D-Fender: what do you play in the band? |
| 15:33.07 | [TK]D-Fender | riddlebox: Guitar, keyboards, vocals (soon with them, plenty on my own). Bass on my ownwhenever its needed... |
| 15:33.38 | riddlebox | dang a one man band |
| 15:34.08 | [TK]D-Fender | riddlebox: I don't do drums. That kind of coordinate would take a LOT of work. |
| 15:35.01 | riddlebox | yeah i play guitar errr used to anyway havent touched it in a bit |
| 15:39.17 | riddlebox | [TK]D-Fender: we have a customer that wants to put in an asterisk system but they want to be able to make calls to their IP Office and I got a SIP trunk working between my test systems but cannot get to the pstn from asterisk through the ip office |
| 15:39.57 | [TK]D-Fender | riddlebox: Time to call Avaya support :) |
| 15:40.15 | [TK]D-Fender | riddlebox: If the call arrives and it does go where you want it to... that's THEIR problem :) |
| 15:40.28 | riddlebox | yeah you know how much they charge an hour now |
| 15:40.33 | riddlebox | 600 bucks an hour |
| 15:42.13 | riddlebox | [TK]D-Fender: my test system has the 3rd party sip license on it, so I set up an extension and had asterisk connected to it, I could call to asterisk, but couldnt call out from asterisk, I connected xlite to it, and it worked fine |
| 15:57.13 | troy42 | riddlebox: how's the debugging granularity on the avaya? geared for paying someone $600/hr? :-) |
| 15:59.48 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:06.02 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 16:11.43 | riddlebox | troy42: I can actually see alot, on monitor, its just odd, it works with xlite, but not with asterisk(with asterisk I see the call come in but nothing goes through) |
| 16:12.11 | riddlebox | troy42: http://pastebin.ca/1893624 |
| 16:30.55 | troy42 | interesting |
| 16:33.15 | riddlebox | not sure where asterisk is losing the call, but it is coming into asterisk |
| 16:35.16 | *** join/#asterisk dmz (~dmz@152.sub-97-18-172.myvzw.com) |
| 16:37.50 | *** join/#asterisk jmacz (~jmacz@201.244.171.150) |
| 16:42.19 | pabelanger-lap | riddlebox: Are you behind a NAT? |
| 16:42.58 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 16:58.00 | pepselap | mornin, gents |
| 17:10.08 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 17:11.21 | xheliox | OT: Does anyone happen to know if Teliax or voip.ms supports SIP TLS? |
| 17:14.25 | redax | hi. |
| 17:30.37 | niekie | Hi all. I've just set up chan_mobile (from Asterisk-Addons) in the trunk, and it seems to have succesfully connected to my phone. However, I am getting a *LOT* of Jitter on the call audio. |
| 17:30.43 | niekie | Anybody know why that might be? |
| 17:39.48 | *** join/#asterisk dauergast (~sag@188-193-228-23-dynip.superkabel.de) |
| 17:40.07 | [TK]D-Fender | niekie: Your networking. |
| 17:40.27 | dauergast | Hi, please, how can I disable the display of notices in the CLI? really annoying |
| 17:40.28 | niekie | [TK]D-Fender: I've tried a call directly from the console. |
| 17:40.35 | niekie | [TK]D-Fender: I still get the jitter. |
| 17:42.23 | [TK]D-Fender | niekie: From console to where? who are you talking to to hear this? |
| 17:42.35 | [TK]D-Fender | niekie: Pplease descibe these calls in proper detail |
| 17:45.28 | *** join/#asterisk Alagar (~Administr@122.164.39.233) |
| 17:48.07 | niekie | [TK]D-Fender: Console -> chan_mobile -> Bluetooth adapter -> Android phone -> My mobile carrier's customer service. |
| 17:48.10 | *** join/#asterisk MiserySoft (~MiserySof@host81-148-116-146.in-addr.btopenworld.com) |
| 17:50.01 | niekie | It sounds stuttery. |
| 17:52.08 | [TK]D-Fender | niekie: Congratulations on using the LEAST likely solution to find support.... |
| 17:52.44 | [TK]D-Fender | niekie: Could be BT itself, or your phone. Or your carrier |
| 17:52.55 | [TK]D-Fender | niekie: Try with another phone |
| 17:53.33 | niekie | would try it with another phone, if he had one. |
| 17:54.28 | redax | try without a phone :) |
| 17:55.21 | niekie | Calls that don't go over Bluetooth seem to work fine. |
| 17:55.45 | niekie | I'll try it with a higher powered server in a while, though. I guess this server might be somewhat underpowered :) |
| 17:57.02 | [TK]D-Fender | niekie: What are you using? |
| 17:58.04 | niekie | An old 800MHz spare box. |
| 17:58.31 | [TK]D-Fender | niekie: Should eb more than fine for BT |
| 17:58.44 | niekie | *nods* That's what I thought. |
| 17:58.57 | niekie | I'll give it a try on this 2,2GHz DC one though :) |
| 17:59.22 | niekie | Nothing to lose :) |
| 18:03.09 | Jumpie | lol |
| 18:05.32 | niekie | And I've got enough spare time to try that for now :) |
| 18:05.35 | niekie | *chuckles* |
| 18:07.57 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 18:07.59 | redax | anybody familiar with LCR? |
| 18:11.14 | [TK]D-Fender | Which? |
| 18:17.42 | redax | linux call router.. |
| 18:28.40 | *** join/#asterisk sarah_p (~sarah_p@pool-71-117-252-127.ptldor.fios.verizon.net) |
| 18:34.41 | *** join/#asterisk MiserySoft (~MiserySof@host81-148-116-146.in-addr.btopenworld.com) |
| 18:36.50 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 18:40.23 | Alton35 | You a fellow Ruger fan? |
| 18:44.30 | niekie | Hmph, no go, still broken. |
| 18:45.13 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 18:46.27 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 18:48.06 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 18:48.06 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 18:50.59 | niekie | Going to try an incoming call. |
| 18:52.19 | riddlebox | pabelanger-lap: yes I am behind nat, but both systems are on the same subnet actually right next to each other |
| 18:54.34 | pabelanger-lap | riddlebox: No need to NAT calls on the same localnet. |
| 19:09.08 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 19:09.09 | [sr] | :) |
| 19:20.16 | riddlebox | pabelanger-lap: I have nat off on the trunk, I may just try h323 and see how well that works |
| 19:20.50 | riddlebox | pabelanger-lap: I am really just behind a router/firewall |
| 19:26.58 | *** join/#asterisk wierdo (~jimmy@77.78.3.197) |
| 19:29.40 | *** join/#asterisk fcolista (~quassel@151.65.93.205) |
| 20:02.56 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 20:06.55 | *** join/#asterisk cesar_CR (~cesar@201.199.168.170) |
| 20:07.25 | *** join/#asterisk cesar_CR (~cesar@201.199.168.170) |
| 20:12.38 | [sr] | people |
| 20:12.52 | [sr] | can someone take a look at this?: http://www.3cx.com/forums/dtmf-tone-is-not-submitted-correctly-including-log-13711.html |
| 20:13.04 | [sr] | i'm not sure if the problem is on asterisk on this softphone |
| 20:13.22 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 20:15.08 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 20:15.52 | pabelanger-lap | [sr]: Have your looked at the RTP packages? |
| 20:16.15 | pabelanger-lap | s/packages/packets/ |
| 20:17.19 | [sr] | pabelanger-lap: well no, i'm not into this... for now, but i chose only SIPINFO DMTF and now it works OK |
| 20:17.38 | [sr] | the other methods may be missing in asterisk i guess |
| 20:18.30 | pabelanger-lap | [sr]: with SIP you want rfc2833 for DTMF. If you have problems, enable RTP debugs and check out the packets |
| 20:19.15 | [sr] | RFC2833 has several options here |
| 20:19.25 | [sr] | a number from 96 to 126 |
| 20:19.33 | [sr] | which one should i chose? |
| 20:19.35 | thevoke | win 49 |
| 20:19.56 | pabelanger-lap | [sr]: options where? |
| 20:20.04 | [sr] | on this softphone |
| 20:21.24 | pabelanger-lap | [sr]: for payloads? |
| 20:21.50 | [sr] | yap, that |
| 20:22.49 | pabelanger-lap | [sr]: you shouldn't need to set them, your phone _should_ change them depending on the event |
| 20:23.05 | [sr] | hum oki... |
| 20:23.23 | [sr] | i just set SIPINFO DMTF and works ok now.. |
| 20:24.04 | pabelanger-lap | [sr]: 101 should be the default payload for dtmf |
| 20:24.15 | [sr] | correct :) |
| 20:24.20 | [sr] | that's what is set |
| 20:26.15 | [sr] | ops.. i have a voicemail message with 74921 seconds... |
| 20:37.38 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 20:39.10 | [sr] | guy |
| 20:39.12 | [sr] | guys |
| 20:39.18 | [sr] | there's a brokenm link on the asterisk page |
| 20:39.23 | [sr] | http://www.asterisk.org/downloads |
| 20:39.35 | [sr] | "don't fear the fax" |
| 20:39.41 | [sr] | on the "download and install", it's broken |
| 20:43.07 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 20:49.21 | [sr] | pabelanger-lap: whos the webmaster? to let him know this |
| 20:50.13 | pabelanger-lap | [sr]: http://www.asterisk.org/contact |
| 20:52.46 | [sr] | done |
| 20:52.47 | [sr] | :) |
| 20:54.43 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:56.01 | *** join/#asterisk markitoxs (~miranda83@cpc8-sgyl21-0-0-cust21.sgyl.cable.virginmedia.com) |
| 21:00.06 | Mhaddog | probably there doing maintenance... it's a long weekend... in almost the middle of the year... so sounds like planned |
| 21:07.19 | *** part/#asterisk Jomu (~Jomu@188.124.200.2) |
| 21:11.24 | Crymsonite | If I reload iax.conf, I should get some feedback that it reloaded right? |
| 21:11.46 | Crymsonite | iax2 reload |
| 21:13.15 | p3nguin_ | not unless you have turned up the verbose level. |
| 21:14.20 | Crymsonite | Usually 3 is good enough. It shows other reloads. |
| 21:14.32 | Crymsonite | I'll set it higher |
| 21:14.44 | p3nguin_ | 3 should be enough. |
| 21:14.55 | Docteh | anyone here have a retail copy of eyeBeam 1.5? I'm wondering if it has G.722 |
| 21:17.19 | Crymsonite | I set it to 9, I am not getting any errors, but I am also not getting any feedback that it's reloaded. It has to do something with my iax.conf, i know it. |
| 21:17.38 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 21:25.15 | Crymsonite | Figured it out, thanks. |
| 21:27.46 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 21:27.53 | [sr] | going to sleep |
| 21:28.48 | *** join/#asterisk fcolista (~quassel@151.65.93.205) |
| 21:39.52 | *** join/#asterisk fcolista (~quassel@151.65.93.205) |
| 21:40.56 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 22:10.46 | *** join/#asterisk sarah_p (~sarah_p@pool-71-117-252-127.ptldor.fios.verizon.net) |
| 22:15.36 | *** join/#asterisk Beirdo (~gjhurlbu@unaffiliated/beirdo) |
| 22:31.13 | ruben23 | hi how do i restart and start dahdi on centos distro..? is it service dahdi restart..? |
| 22:37.54 | *** join/#asterisk Beirdo (~gjhurlbu@unaffiliated/beirdo) |
| 22:50.12 | *** join/#asterisk mrchrisadams (~Adium@CPE-138-130-103-47.lns2.cht.bigpond.net.au) |
| 22:51.31 | *** join/#asterisk dauergast (~sag@188-193-228-23-dynip.superkabel.de) |
| 22:55.10 | dauergast | hi, is it possible to use wildcards within variables? something like this: gotoif $[${callerid(num)} = "_**." ]?s,1;s,4 ? |
| 23:02.19 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 23:23.36 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:24.28 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 23:28.51 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 23:29.06 | *** join/#asterisk Entulho (~foo@189.31.112.46) |
| 23:31.05 | *** join/#asterisk sarah_p (~sarah_p@pool-71-117-252-127.ptldor.fios.verizon.net) |
| 23:37.46 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:59.45 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |