| 00:01.11 | [TK]D-Fender | \o/ |
| 00:01.26 | [TK]D-Fender | Qsynth + JACK + MIDI = FTMFW! |
| 00:01.39 | [TK]D-Fender | can officially dump Windows |
| 00:01.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:01.57 | [TK]D-Fender | jaytee: ! |
| 00:02.00 | beek | [TK]D-Fender: Congrats! |
| 00:02.11 | [TK]D-Fender | Qsynth + JACK + MIDI = FTMFW![19:01]<[TK]D-Fender>\o/ |
| 00:02.13 | [TK]D-Fender | [19:01]<[TK]D-Fender>Qsynth + JACK + MIDI = FTMFW! |
| 00:02.13 | jaytee | [TK]D-Fender,!!!! |
| 00:02.17 | jaytee | yay!!! |
| 00:02.21 | [TK]D-Fender | [19:01]* [TK]D-Fendercan officially dump Windows |
| 00:02.26 | jaytee | awesome |
| 00:02.35 | [TK]D-Fender | jaytee: Victory is mine! |
| 00:02.54 | jaytee | hands [TK]D-Fender a Cohiba |
| 00:02.55 | [TK]D-Fender | jaytee: Camera & printer both work just fine, so I'm good to go... |
| 00:03.37 | [TK]D-Fender | jaytee: Just did some playing around in /dev and saw the name the drive gave it, did some choosing between OSS vs ALSA, etc, and bam works like a charm |
| 00:04.04 | jaytee | [TK]D-Fender, my whole day went to shit. my new ivr server is fubared, I couldn't get the lumenvox 8.6.8 engine working right and it started screwing up asterisk. zaptel won't load at boot anymore. |
| 00:04.26 | [TK]D-Fender | jaytee: :/ |
| 00:04.47 | jaytee | everything was working fine until I installed Lumenvox. of course they claim they've never seen this problem. |
| 00:05.10 | *** join/#asterisk saftsack (n=oliver@g226135248.adsl.alicedsl.de) |
| 00:05.35 | beek | jaytee: Test box or production? |
| 00:05.40 | jaytee | so now I'm at home and VPN'd into work with an SSH connection to the box still trying to rework stuff. |
| 00:05.51 | jaytee | beek, this is slated to be the production box. |
| 00:06.08 | beek | jaytee: Many hours of lost time I assume...? |
| 00:06.30 | jaytee | about 3/4 of my day so far |
| 00:06.51 | Qwell | [TK]D-Fender: rosegarden? |
| 00:06.57 | beek | Shit. Well, believe it or not my PRI problem has FINALLY been solved as of last Friday. Not bad -- only six weeks to repair. |
| 00:07.27 | beek | wipes up the sarcasm his last comment dripped. |
| 00:07.54 | Qwell | beek: is it from a big telco? |
| 00:07.56 | [TK]D-Fender | Qwell: Qsynth. Its a GTK front end to fluidsynth which is a SoundFont (creative sound bank format) soft-synth. Basically lets me play my dumb MIDI controller "live" with sounds |
| 00:08.03 | beek | Qwell: Level 3 |
| 00:08.21 | Qwell | [TK]D-Fender: I know what qsynth is. :D check out rosegarden |
| 00:08.34 | [TK]D-Fender | Qwell: `Already installed ;) |
| 00:08.41 | [TK]D-Fender | Qwell: Along with Timidity, Ardour, etc :D |
| 00:08.50 | Qwell | (last time I used it, it was rather buggy and would hang the system a lot) |
| 00:09.07 | Qwell | pretty cool though |
| 00:09.13 | [TK]D-Fender | Qwell: Rosegarden saw my keyboard first, but its complex shit! Couldn't find out how to go into live" mode with a sound bank |
| 00:09.19 | [TK]D-Fender | Qwell: I just wanna f-ing PLAY! |
| 00:09.47 | [TK]D-Fender | Qwell: Most of the time I jsut sit on my one good grand piano SF and play |
| 00:11.09 | jaytee | what would prevent zaptel from starting as a service (it did before today) but typing service zaptel start starts it fine from the command line? |
| 00:11.35 | *** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 00:11.44 | beek | jaytee: chkconfig --list zaptel |
| 00:12.06 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-1839f9f224df9648) |
| 00:12.35 | jaytee | beek, it lists runlevels 0, 1 and 6 as off and 2 through 5 as on |
| 00:13.16 | beek | Of course you're not at the console... I'd be interested in seeing what puked during the startup. This is CentOS 5, right? |
| 00:13.55 | jaytee | beek, yes. I'm not at the console, I'm running Putty over a VPN |
| 00:14.28 | beek | jaytee: For fun, type: chkconfig zaptel off, then chkconfig zaptel on |
| 00:15.24 | jaytee | beek, ok. did that and rebooted, waiting for it to come back online |
| 00:15.52 | beek | jaytee: Once it's back online look in /var/log/messages |
| 00:16.07 | *** join/#asterisk marv0997 (n=marv@205.211.247.62) |
| 00:16.21 | *** part/#asterisk marv0997 (n=marv@205.211.247.62) |
| 00:16.31 | jaytee | beek, will do. |
| 00:20.34 | jaytee | beek, this time in messages I saw the zaptel modules load ok and when I run an asterisk remote console and type help my zap commands are all there. looks like toggling the chkconfig bit did the trick. |
| 00:20.46 | beek | jaytee: Cool. |
| 00:20.58 | ph8 | does anyone know if i can get my GXP2000 (Grandstream) to connect via SIP outside the university's anal firewall? I've got a dynamic SSH Tunnel setup as a SOCKS4 proxy - can i tell it to use that somehow? |
| 00:22.06 | jaytee | now if could only figure out how to change my repo config to download 8.5 instead of 8.6 of Lumenvox. |
| 00:22.31 | beek | svn? |
| 00:22.56 | jaytee | no, i can either use yum or I can download the rpms and run rpm from the command line |
| 00:23.12 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 00:23.28 | jaytee | brb |
| 00:23.43 | lowtek | ph8: afaik there's no way to use a socks proxy and SIP/RTP. |
| 00:24.03 | ph8 | hi lowtek |
| 00:24.04 | ph8 | rtp? |
| 00:24.04 | lowtek | ph8: SIP just sets up and manages the call, the audio goes via RTP. |
| 00:24.12 | ph8 | :o |
| 00:24.19 | ph8 | should i not be able to just forward the right ports? |
| 00:24.25 | ph8 | i've been thinking i'm just missing a couple? |
| 00:24.34 | lowtek | 5060, 20000-40000 |
| 00:24.40 | ph8 | eep to the latter |
| 00:25.01 | ph8 | can i not restrict ports to say, 10? |
| 00:26.20 | lowtek | ph8: I know you can specify the port range with rtp.conf on the asterisk side. Don't know how many RTP ports a single call needs though. Sorry, I was wrong, it's 10000-20000 by default. |
| 00:26.53 | lowtek | lowtek: Easier solution would be to just do an any-any rule to your asterisk IP and then set your peers to nat=yes in sip.conf |
| 00:27.27 | ph8 | it's the any-any rule that's the issue? |
| 00:27.30 | ph8 | the phone is connected to a router |
| 00:27.34 | lowtek | lol, talking to myself, that was to you ph7 |
| 00:27.36 | lowtek | ph8 |
| 00:27.37 | lowtek | dangit |
| 00:27.38 | ph8 | which is connected to the network |
| 00:28.02 | ph8 | so i could perhaps tell the phone that my PC (192.168.1.10) is its SIP server |
| 00:28.02 | lowtek | ph8: couldn't tell you what your issue is without intimate details of all equipment involved. |
| 00:28.15 | ph8 | and have my PC forward (via SSH tunnels) the appropriate ports |
| 00:28.53 | ph8 | phone + pc into router -> uni network (anal firewall, not packet shaping just port blocking) -> publicly exposed server |
| 00:28.58 | lowtek | ph8: Don't think that will work very well. You can try a simple ipsec tunnel, they work pretty well with sip/rtp. |
| 00:29.16 | ph8 | sounds interesting |
| 00:29.31 | ph8 | i could generate a script that just generates rules for 5060 + about 500 rtp ports or something |
| 00:29.38 | ph8 | is RTP likely to use more than one port per call? |
| 00:29.52 | lowtek | ph8: lol, or just use a cell phone. That seems like an awful lot of craziness for a single phone. Oh,and grandstream sucks, throw that thing away. |
| 00:30.10 | ph8 | well, calls in to the grandstream would be free |
| 00:30.20 | ph8 | calls into the cell incur outbound trunk charges |
| 00:30.44 | lowtek | ph8: I guess it depends on what your time is worth. |
| 00:30.51 | ph8 | :p |
| 00:31.06 | ph8 | i don't get that many calls, i just don't like having hardware that doesn't work on my desk |
| 00:32.58 | ph8 | grandstream is shit also, who should i buy from? |
| 00:33.06 | lowtek | Polycom. |
| 00:33.14 | ph8 | will they let me route everything through a socks proxy? :p |
| 00:33.35 | lowtek | No. |
| 00:33.54 | lowtek | You're shit out of luck with the ssh tunnels and socks proxies. You can try a IPSEC tunnel, it's encapsulating. |
| 00:34.07 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 00:34.19 | ph8 | :o what doesn't the ssh tunnel encapsulate? |
| 00:34.23 | ph8 | udp stuff? |
| 00:34.29 | lowtek | ph8: everything |
| 00:50.53 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 00:51.53 | beek | jaytee: I'm heading out for the night. As for your question about getting a specific version from the repo, do a 'man yum' and look under MISC/Specifying package names |
| 00:51.56 | beek | GN all |
| 00:52.05 | jaytee | nite |
| 00:53.14 | *** part/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 01:05.42 | *** join/#asterisk riddlebox (n=user@75-132-195-207.dhcp.stls.mo.charter.com) |
| 01:06.15 | *** join/#asterisk makman111 (n=jmaki@75-168-212-143.mpls.qwest.net) |
| 01:06.28 | makman111 | Anyone out there willing to help a newbie out? |
| 01:07.11 | [TK]D-Fender | ~ask |
| 01:07.12 | jbot | somebody said ask was Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:07.12 | makman111 | Need to set up a *now server. Have DID from broadband.com |
| 01:10.10 | jaytee | aaahhhhh, the sweet feeling of success!!!! |
| 01:16.44 | *** join/#asterisk qdk (n=qdk@79.138.248.33.bredband.3.dk) |
| 01:20.02 | *** join/#asterisk Siya (n=djerk@suzy.djerk.nl) |
| 01:28.18 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 01:34.03 | aptura | which asterisk version uses core in its cli? 1.6? |
| 01:34.18 | lowtek | 1.4 + |
| 01:34.43 | aptura | for some reason its not reading my existing conf files |
| 01:35.29 | Micc | anyone know much about shoretel pbx systems? |
| 01:35.56 | lowtek | Micc: No, this is #asterisk |
| 01:36.41 | Micc | lowtek, I know but my next question is, how do I get it to work with asterisk? |
| 01:37.03 | lowtek | lowtek: What are you trying to do exactly with all the details? |
| 01:37.36 | lowtek | Ugh, talking to myself again. I really need to lay off the smack. That was to you, Micc. |
| 01:38.23 | *** join/#asterisk mog (n=mog@c-68-62-217-121.hsd1.al.comcast.net) |
| 01:38.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:39.06 | Micc | I have a customer that currently uses some shoretel equipment. I wand to switch them over to using SIP lines. |
| 01:39.28 | lowtek | Ahh. I'll help for $125/hr. |
| 01:39.58 | jaytee | lol |
| 01:40.11 | Micc | lowtek, will it only take one hour? |
| 01:40.18 | *** join/#asterisk M-33 (n=meow@67.159.178.20) |
| 01:40.23 | lowtek | No, minimum 3 hours. |
| 01:40.31 | lowtek | Interested? |
| 01:40.43 | M-33 | hello, by default installation, does the three way calling enabled? |
| 01:40.46 | jaytee | hehehe |
| 01:40.49 | Micc | Maybe, if you teach me how its done at the same time. |
| 01:41.02 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 01:41.03 | lowtek | Not really, just messing with you. I found it funny that you're in here trying to get free help for one of your paying customers on a phone system that's not asterisk. |
| 01:41.34 | jaytee | played like a Spinnet piano :-) |
| 01:41.36 | lowtek | But, iirc, shortel will do sip natively. You can use Polycom phones with their equipment and they are SIP. |
| 01:42.47 | Micc | yeah I saw that. |
| 01:43.02 | Micc | I just read an article that says it has SIP trunking. |
| 01:43.15 | Micc | So I imagine its got to be pretty easy to setup. |
| 01:44.12 | lowtek | Maybe call shortel? |
| 01:48.46 | aptura | sip history |
| 01:49.55 | aptura | stop now |
| 01:50.09 | aptura | hay i have a questions about some of the core commands |
| 01:50.35 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 01:50.36 | aptura | seems thay are not responding |
| 01:50.42 | aptura | at least some of them |
| 01:51.15 | aptura | http://solutionsathand.files.wordpress.com/2007/05/asteriskcli.txt |
| 01:52.21 | aptura | cli responds to core but not sip history |
| 01:54.33 | aptura | guess its the way its layed out. |
| 01:56.05 | aptura | Micc |
| 01:56.14 | Micc | yeah |
| 01:56.35 | aptura | does this look normal? This is why it is not reading my extentions.conf |
| 01:56.39 | aptura | extconfig /etc/asterisk/extconfig.conf |
| 01:57.34 | aptura | did a config list and thats what cli came up with.It is not obvios why my cli is not responding to my old extentions.conf file |
| 01:57.50 | aptura | not/now |
| 01:58.34 | Micc | how did you do a config list? |
| 01:59.01 | Micc | I don't seem to have a "config list" command in 1.4 |
| 01:59.01 | aptura | just typed in config list on CLI |
| 01:59.25 | Micc | I don't have that. |
| 01:59.28 | aptura | mmmm |
| 01:59.52 | aptura | this is 1.4.22 but do not think the changes would make a difference. |
| 02:02.04 | *** join/#asterisk rcy` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 02:03.14 | *** join/#asterisk Avelino (n=Avelino@189-46-71-203.dsl.telesp.net.br) |
| 02:06.10 | aptura | call it a night |
| 02:08.24 | Micc | Anyone know a manufacturer of small low cost asterisk devices? someone told me about them once. |
| 02:11.16 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:11.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:14.13 | voxter | Micc: pika technologies, aastra telecom, are a couple |
| 02:15.07 | *** join/#asterisk etfonhomey (n=chatzill@74-131-86-46.dhcp.insightbb.com) |
| 02:15.32 | *** join/#asterisk Aptura (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:15.42 | [TK]D-Fender | What is an "asterisk device" |
| 02:15.46 | [TK]D-Fender | a SIP phone? |
| 02:15.50 | [TK]D-Fender | a TDM card? |
| 02:16.06 | [TK]D-Fender | A hard drive? * is softwar.e.... software sits on media... |
| 02:16.11 | [TK]D-Fender | maybe a USB key? |
| 02:16.20 | [TK]D-Fender | I know.. |
| 02:16.31 | [TK]D-Fender | pulls out a 5 1/4" floppy! |
| 02:16.39 | [TK]D-Fender | Micc: Here you go! |
| 02:18.14 | Micc | gee, thanks, I think. |
| 02:21.41 | [TK]D-Fender | Micc: Maybe... just MAYBE you should be a bit more specific in your requests ;) |
| 02:23.13 | Micc | I think it was hardwire, who told me about his company working on a little box that could handle about 10 phones. |
| 02:23.57 | [TK]D-Fender | Micc: Plenty of embedded devices can handle that. |
| 02:24.16 | [TK]D-Fender | Micc: Soekris, PCEngines, Pika is a dedicated one, lost of pothers |
| 02:24.18 | [TK]D-Fender | others* |
| 02:25.08 | Aptura | <PROTECTED> |
| 02:25.42 | Aptura | I think this is causing some issues of the phones extentions not being read properyly |
| 02:25.59 | Aptura | Any comments? |
| 02:27.17 | *** join/#asterisk etfonhomey (n=chatzill@74-131-86-46.dhcp.insightbb.com) |
| 02:29.00 | [TK]D-Fender | Aptura: phones extensions? huh? |
| 02:29.06 | [TK]D-Fender | Aptura: Details would help |
| 02:32.24 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:33.15 | Aptura | sorry my system is acting a tad flaky |
| 02:38.25 | jaytee | like a freshly baked croissant |
| 02:38.33 | NovceGuru | mmmm |
| 02:38.41 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 02:40.05 | Aptura | sorry TK, did not have the details. Put it in pastebin. http://www.pastebin.ca/1307006 |
| 02:41.09 | [TK]D-Fender | Aptura: Nifty... now actually LOOK at the call. |
| 02:42.03 | Aptura | This was a fully working system untill one day was playing with Dial command on cli then got some errors. |
| 02:42.34 | Aptura | it would dial the extention and goto vm but got some errors and soon after no phone would respond dialingout. |
| 02:42.55 | Aptura | thats not the entire dial plan |
| 02:42.57 | [TK]D-Fender | Aptura: and none of that matters an ounce until you actually look at a failed call in DETAIL |
| 02:44.32 | Aptura | chan_sip.c:14383 handle_request_invite: Call from '200' to extension '8500' rejected because extension not found. |
| 02:44.36 | Aptura | you mean that? |
| 02:45.06 | [TK]D-Fender | Aptura: I mean that isn't SIP DEBUG and you're not looking at whats REALLY going on. |
| 02:45.17 | Aptura | I see |
| 02:45.28 | *** join/#asterisk dandate2 (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net) |
| 02:45.36 | dandate2 | woohoo i got it |
| 02:45.39 | dandate2 | it works! |
| 02:45.55 | jaytee | it does? what is "it"? |
| 02:46.52 | dandate2 | my pbx server |
| 02:46.53 | dandate2 | 12016206323 |
| 02:47.00 | dandate2 | 1-201-620-6323 |
| 02:47.04 | [TK]D-Fender | OU812? |
| 02:47.16 | SlicerDicer | [TK]D-Fender: I am getting a 57i :) |
| 02:47.38 | [TK]D-Fender | SlicerDicer: Bleh... Mine made me wish for my old bed-side Polycom IP301 |
| 02:47.49 | dandate2 | 3 |
| 02:47.51 | SlicerDicer | [TK]D-Fender: its under 100$ |
| 02:47.53 | SlicerDicer | cant complain |
| 02:49.18 | dandate2 | ?? softphone |
| 02:50.51 | SlicerDicer | [TK]D-Fender: my 480i should be here tomorrow (yay) |
| 02:50.57 | Aptura | http://www.pastebin.ca/1307011 here is stip debug. Do not have to much experaince reading it |
| 02:52.54 | [TK]D-Fender | Aptura: PB "dialplan show" |
| 02:52.59 | Aptura | k |
| 02:53.11 | [TK]D-Fender | SlicerDicer: 480i is a more solid phone. |
| 02:56.45 | Micc | What is the difference between an analog line and an analog trunk? |
| 02:57.10 | Micc | When it says supports analog trunking, what does that mean? |
| 02:58.11 | [TK]D-Fender | Micc: Means "I can't pick consistent standard terminology" |
| 02:58.33 | [TK]D-Fender | Micc: Doesn't imply HOW it supports "analog" |
| 03:00.03 | Aptura | http://www.pastebin.ca/1307016 |
| 03:00.51 | Micc | ok. :) |
| 03:01.10 | Aptura | mmm |
| 03:01.37 | [TK]D-Fender | Aptura: Yup, I see NONE of what you showed earlier |
| 03:01.38 | *** join/#asterisk joako (n=joako@adsl-9-115-80.mia.bellsouth.net) |
| 03:01.55 | [TK]D-Fender | Aptura: Now CAT it from CLI and show me your entire /etc/asterisk/ folder |
| 03:02.12 | [TK]D-Fender | (ls -la /etc/asterisk) |
| 03:02.18 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 03:02.24 | Aptura | okay |
| 03:03.14 | joako | I am trying to figure out why I can not transfer SIP calls. I am using Asterisk 1.4.22. It does not matter what phone I use, Linksys, Grandstream or Polycom... blind or attended transfer does not work |
| 03:03.26 | joako | Does anyone have an idea where to start checking? |
| 03:04.13 | Aptura | http://www.pastebin.ca/1307019 |
| 03:05.09 | Aptura | jako, let me see your extention for one phone in extentions.conf you probebly did not inclute the t switch |
| 03:06.28 | Aptura | example of this extention exten => 200,1,Dial(SIP/${EXTEN},20,Tt) |
| 03:06.28 | Aptura | exten => 200,2,VoiceMail(u200@default) |
| 03:06.28 | Aptura | exten => 200,102,VoiceMail(b2009@default) |
| 03:06.28 | Aptura | exten => 200,103,Hangup() |
| 03:07.02 | [TK]D-Fender | Aptura: that isn't SIP transfer.. |
| 03:07.02 | Aptura | notice on the end Tt |
| 03:07.22 | [TK]D-Fender | Aptura: thats * features.conf based transfer, not a SIP transfer |
| 03:07.33 | Aptura | okay well then some thing else |
| 03:07.40 | [TK]D-Fender | Aptura: issue a reload and watch for the load. |
| 03:08.08 | [TK]D-Fender | Aptura: If you don't see it loading your dialplan, pastebin your modules.conf |
| 03:08.19 | joako | Aptura: I am not (nor do I desire to) use that sort of transfer. I am using SIP tranfer which is handset dependent.... I see the phone sending a SIP REFER request and Asterisk answers back "SIP/2.0 603 Declined (policy)" |
| 03:09.25 | [TK]D-Fender | joako: PB full SIP debug of a call and your sip.conf |
| 03:10.25 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 03:11.05 | rickross | which one is actually less likely to segfault? 1.6.0.3 or 1.6.1b4 ? |
| 03:11.08 | Aptura | TK, you did not mention what to look for in the reaload. |
| 03:12.25 | [TK]D-Fender | Aptura: watching your dialplan get loaded |
| 03:12.36 | Aptura | I just reloaded it |
| 03:12.50 | [TK]D-Fender | hands rickross a tourniquette |
| 03:13.09 | rickross | TK - thanks for that :) |
| 03:13.24 | rickross | is 1.6 just a disaster? |
| 03:14.03 | rickross | We had tried it a while back, but there were too many niggling issues, so we thought maybe by now it would have stabilized somewhat |
| 03:14.16 | rickross | I guess I was being too optimistic? |
| 03:14.40 | [TK]D-Fender | rickross: 1.6.1 is still beta... |
| 03:15.00 | [TK]D-Fender | rickross: 1.6.0 is still somewhat new, but should statistically be more stable... |
| 03:15.10 | rickross | 1.6.0.3 isn't, and it segfaulted within minutes of being started up on our machine |
| 03:15.22 | [TK]D-Fender | rickand 1.6.0.2? |
| 03:15.37 | jaytee | or try 1.6.0.1 |
| 03:15.39 | rickross | the beta notes claim to address some crashes, so I had fingers crossed |
| 03:16.19 | rickross | anyone hererunning 1.6.x for production? |
| 03:16.45 | joako | [TK]D-Fender: I'm here... working on sanitizing what you are requesting |
| 03:16.49 | [TK]D-Fender | Aptura: And? If no go, PB modeules.conf as I asked |
| 03:17.24 | Aptura | okay |
| 03:19.01 | Aptura | chan_dahdi.so is aleady loaded |
| 03:19.06 | Aptura | http://www.pastebin.ca/1307027 |
| 03:19.15 | Aptura | kind of checked it to be sure |
| 03:20.04 | [TK]D-Fender | Aptura: dahdi has nothing to do with this |
| 03:20.18 | joako | [TK]D-Fender: In the process of sanitizing my sip.conf i saw allowtransfer=no, setting allowtransfer=yes fixed my issue. Thanks! |
| 03:20.23 | Aptura | k |
| 03:20.24 | [TK]D-Fender | Aptura: try "module load pbx_config.so |
| 03:20.47 | [TK]D-Fender | joako: That'll learn ya... |
| 03:21.47 | Micc | So what is an ITSP trunk? |
| 03:22.03 | [TK]D-Fender | Micc: What do you keep finding these retard terms? |
| 03:22.11 | [TK]D-Fender | ~itsp |
| 03:22.12 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:22.21 | Micc | oh, ok. |
| 03:22.27 | [TK]D-Fender | Micc: Nice ot specify even a PROTOCOL |
| 03:23.01 | Micc | Yeah, they don't mention a protocol, just itsp. |
| 03:23.07 | Micc | shoretel website. |
| 03:23.09 | [TK]D-Fender | Micc: "they"? |
| 03:23.26 | [TK]D-Fender | Micc: And the reason you're even looking at them? |
| 03:23.55 | Aptura | ack, found the issue. some text some how was typed in perhaps by accident on extentions.conf |
| 03:24.08 | Aptura | so now that model is loaded and its up and running again. |
| 03:24.13 | Aptura | module |
| 03:24.24 | Micc | Because one of my customers has it and I need to see if we'll be able to give them some SIP lines. |
| 03:24.55 | [TK]D-Fender | Micc: Not understanding the terminology yourself makes me fear for your customers... |
| 03:24.57 | Aptura | it was on the first line on extentions.conf. I deleted it so its running. There are some errors but unrelated to this one. |
| 03:25.55 | [TK]D-Fender | Aptura: Something we might have seen if you didn't truncate things at the start.... |
| 03:26.19 | Aptura | TK, do you think that is why perhaps alot of these mom and pop asterisk companies went out of business or just to much compitition? |
| 03:28.08 | [TK]D-Fender | Aptura: What is why? |
| 03:30.57 | Aptura | Well since this business is so critical that perhaps admins do not know as much about the fine details of asterisk and they could have unresolved or delayed issues causing there customers to leave. I have heard the complaints from people who were the customers of other pbx sip based services and left. The state of the economy does not help either in this case. |
| 03:31.22 | Aptura | It is just what I have observed. |
| 03:32.28 | [TK]D-Fender | Aptura: Incompetent admins can sink any project. Phones are just something that business owners don't want to have to worry about that much. |
| 03:32.43 | *** join/#asterisk Blackthorn (n=support@76-77-161-241.smyth.net) |
| 03:33.20 | Blackthorn | Is there a card perhaps by Digium that would support 12 channels of voice and 12 channel v.90 modem? (incoming?) |
| 03:33.41 | *** join/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net) |
| 03:33.46 | [TK]D-Fender | Blackthorn: * does voice, not data |
| 03:34.02 | Aptura | true |
| 03:34.25 | Aptura | http://www.myvoipprovider.com/VoIP_Provider_Graveyard this list is perhaps dated so is probebly alot longer. |
| 03:34.51 | *** join/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com) |
| 03:35.01 | *** join/#asterisk mtutaj (n=mtutaj@76-231-68-228.lightspeed.cicril.sbcglobal.net) |
| 03:35.12 | *** part/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net) |
| 03:35.14 | *** join/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net) |
| 03:35.40 | Sargun | ~onjoin Sargun die |
| 03:35.40 | jbot | Sargun: ok |
| 03:36.03 | mtutaj | I am having an issue accepting incoming calls, I can make outgoing np, not sure where to look |
| 03:36.18 | Aptura | I should make a recover script so I dont make such a dumb mistake as that one. I was perhaps to fast at the keyboard and did not know I pasted something in the extentions.conf file that I would have otherwise known of ;) |
| 03:36.56 | [TK]D-Fender | mtutaj: * CLI <- |
| 03:37.16 | mtutaj | k |
| 03:37.22 | [TK]D-Fender | Aptura: First thing you should fix is looking at little bits of things instead of the whoel picture |
| 03:38.15 | Aptura | true |
| 03:39.25 | Aptura | btw, no one in asterisk forms never replyed with a result on this issue. |
| 03:40.19 | Aptura | anyway, do thank you very much saved me alot of headaches:) |
| 03:40.19 | [TK]D-Fender | Aptura: Well WE never got to see the whole file or your attempts to manually load the module which would have tipped you off... |
| 03:40.27 | [TK]D-Fender | Aptura: So do do yourself a favour and NEVER cut corners in debugging again |
| 03:42.21 | Gopher_77 | I'm trying to install * on netbsd. I'm having trouble compiling the driver. Can I get some assistance? |
| 03:43.18 | NovceGuru | what driver |
| 03:43.31 | Gopher_77 | zaptel |
| 03:43.50 | Gopher_77 | there's something about atari/stand/edahdi in the kernel source though |
| 03:44.01 | Gopher_77 | but the driver isn't loading for my hardware |
| 03:47.29 | Gopher_77 | pcictl shows 000:11:0: unknown vendor 0xe159 product 0x0001 (miscellaneous network) for my 4-port card |
| 03:51.55 | Kobaz | hmmm |
| 03:53.09 | Kobaz | how could i run arbitrary dialplan code on a particular channel |
| 03:53.32 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 03:58.38 | jaytee | Arbitrary Ar"bi*tra*ry, a. [L. arbitrarius, fr. arbiter: cf. |
| 03:58.38 | jaytee | <PROTECTED> |
| 03:58.39 | jaytee | <PROTECTED> |
| 03:58.39 | jaytee | <PROTECTED> |
| 03:58.39 | jaytee | <PROTECTED> |
| 04:00.57 | [TK]D-Fender | Kobaz: load res_chaostheory.so |
| 04:02.11 | jaytee | or you can arbitrarily insert some arbitrary value in ${EXTEN} and arbitrarily Dial($EXTEN}) from an arbitrary location |
| 04:02.24 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 04:03.01 | Blackthorn | lol |
| 04:03.05 | Kobaz | what? |
| 04:03.11 | [TK]D-Fender | Kobaz: X = Y maybe Z <- ILLOGICAL operators. Thats what you need to use... |
| 04:03.12 | jaytee | exactly |
| 04:03.12 | Kobaz | it wasn |
| 04:03.22 | Kobaz | it wasn't that complex of a question |
| 04:03.30 | Kobaz | how about |
| 04:03.38 | jaytee | no, not complex, just vague as all hell |
| 04:03.39 | Kobaz | can i run specific code, on a specific channel |
| 04:03.45 | *** join/#asterisk Jollyr01staup (n=schapman@ip24-255-111-59.dc.dc.cox.net) |
| 04:03.59 | jaytee | sure, but there's nothing arbitrary about that |
| 04:04.00 | [TK]D-Fender | Kobaz: How do you run code on a channel? What does that mean? |
| 04:04.10 | Kobaz | like.... i want Park() to run on channel SIP/23423847-2234sdfasdf34 |
| 04:04.18 | [TK]D-Fender | Kobaz: what does it do to the channel? What is the channel doing while this OTHER stuff is happening? |
| 04:04.22 | Kobaz | dialplay code |
| 04:04.38 | NovceGuru | presence is so cool! |
| 04:04.41 | Kobaz | like you can make a custom feature in features.conf *1234 |
| 04:04.50 | Kobaz | and you can run that on the callee, or the caller channel |
| 04:04.55 | Jollyr01staup | anyone familiar with TE1xx connected to legacy phone system? |
| 04:04.56 | [TK]D-Fender | Kobaz: What syncs actions? Does it even matter? Who shot J.R.? Whats the Caramilk secret? How many angels can dance on the head of a pin? |
| 04:05.32 | [TK]D-Fender | Jollyr01staup: vague as that is, sure... plenty of us. Try asking something more specific now |
| 04:05.44 | Kobaz | so, instead of triggering some code on dtmf |
| 04:05.51 | Kobaz | can i trigger some dialplan code through, say... the ami |
| 04:06.09 | [TK]D-Fender | Kobaz: You can... its called ORIGINATE |
| 04:06.15 | Kobaz | ah |
| 04:06.18 | Kobaz | but... on an existing call |
| 04:06.31 | [TK]D-Fender | Kobaz: Who says it has to be acting upon an established channel? |
| 04:06.37 | Kobaz | oh |
| 04:06.39 | [TK]D-Fender | Kobaz: You are not specifying the ENDS you are trying to meet |
| 04:06.46 | Kobaz | hmm |
| 04:06.52 | [TK]D-Fender | Kobaz: get SPECIFIC. Solutions need to be |
| 04:07.03 | Kobaz | okay, well i guess it's terms |
| 04:07.11 | Kobaz | i want to launch some dialplan code on an established channel |
| 04:07.15 | [TK]D-Fender | Kobaz: We can advise 100 ways that WON'T do what you need if you keep this in "theotretical-land" |
| 04:07.17 | Kobaz | and then go back to the call |
| 04:08.19 | Jollyr01staup | Okay.. I have a TE122 Card that I am connecting to a Comdial phone system. I am using the Dahdi drivers and when I do a pri status in Asterisk I get Status=Provisioned, Down, Active. I am not sure what to set the switchtype too or the signalling too. |
| 04:09.12 | [TK]D-Fender | Jollyr01staup: that would depend on what your PBX is expecting |
| 04:10.23 | Jollyr01staup | I tried several different options and pri_net and pri_cpe seem to give me some response.. but .. I don't even know what is valid for that card to try them all |
| 04:13.07 | [TK]D-Fender | Jollyr01staup: its not "whats valid for the card" its what signalling is YOUR PBX expecting. You don't even have that answer, do you? |
| 04:13.15 | Jollyr01staup | A couple things I do know... it is expecting a T1 24 channel it has many DID's so I know it must be a PRI but thats about it.. |
| 04:13.33 | [TK]D-Fender | PRI has *23* channels |
| 04:13.40 | [TK]D-Fender | and 1 D-chan |
| 04:14.07 | [TK]D-Fender | Jollyr01staup: And nothing implies that that port on your PBX is necessarily PRI. You should know what its configured to |
| 04:16.20 | Jollyr01staup | Okay but for the sake of argument if it is not set to PRI and its a regular T1 card.. what settings would I look to place in the chan_dahdi? |
| 04:16.23 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 04:16.35 | mchou | ~book |
| 04:16.36 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 04:16.59 | [TK]D-Fender | Jollyr01staup: depends if its acting as CPE or NET. Depends on framing, who is providing timing, etc |
| 04:17.50 | dan__t | I wish my Polycom didn't take a day and a half to boot. |
| 04:19.01 | [TK]D-Fender | dan__t: 2 minutes for the rest of us... |
| 04:19.01 | dan__t | Must be nice. |
| 04:19.01 | [TK]D-Fender | dan__t: then again... do it right and you don't NEED to reboot them again |
| 04:19.01 | dan__t | I know it's trying to netboot, suppose I should look in to that. |
| 04:19.01 | dan__t | it reboots itself after a config change. |
| 04:19.08 | wastrel | thx for the help earlier |
| 04:19.09 | *** part/#asterisk wastrel (n=wastrel@nylug/member/wastrel) |
| 04:19.14 | dan__t | Looks like this time, however, it may have froze... |
| 04:19.17 | dan__t | Very nice. |
| 04:20.44 | Jollyr01staup | I cannot tell the phone system has been here for a while and the only way to tell would be to hire someone to come in and link up to it.. thats why I was looking around for example configurations. To see if possibly I could get something to work... |
| 04:20.47 | [TK]D-Fender | danI've only had 2 phones ever freeze on me. 2nd time was today, last was a year ago |
| 04:21.43 | [TK]D-Fender | Jollyr01staup: start as PRI_NET providing timing, then invert. Then switch to FXS_LS then invert on the thought it might be CAS |
| 04:21.52 | [TK]D-Fender | Jollyr01staup: invert timeing on each, etc |
| 04:22.52 | Kobaz | hmm |
| 04:22.54 | Kobaz | ack |
| 04:22.58 | Kobaz | i'm crashing asterisk |
| 04:23.01 | Kobaz | [Jan 12 23:22:11] WARNING[20730]: channel.c:3929 ast_channel_bridge: SIP/5506-08203a80 is already in a bridge with SIP/5501-082079f8 |
| 04:23.04 | Kobaz | [Jan 12 23:22:11] WARNING[20730]: res_features.c:1570 ast_bridge_call: Bridge failed on channels SIP/5506-08203a80 and SIP/5506-081f4038 |
| 04:23.07 | Kobaz | Disconnected from Asterisk server |
| 04:23.13 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:23.38 | Kobaz | 1.4.22 |
| 04:25.19 | dan__t | Yeah its been a while since one of mine has frozen, too. |
| 04:25.23 | Jollyr01staup | when I try FXS or FXO anything... it won't even load the channels in asterisk. |
| 04:26.00 | dan__t | So is there any way to do anything during a Record(), like... put a mark in the recording, or interact with the channel while it's being recorded? What if I wanted to conf someone in while I was in a Record()? |
| 04:27.02 | Jollyr01staup | [TK]D-Fender: this is what asterisk says "Signalling requested on channel 1 is FXO Loopstart but line is in ISDN PRI signalling" |
| 04:27.28 | [TK]D-Fender | Jollyr01staup: means "get consistant in your configs" |
| 04:27.49 | [TK]D-Fender | Jollyr01staup: don't have your system.conf saying one things and your chan_dahdi.conf saying another |
| 04:27.50 | jaytee | nite all |
| 04:28.51 | [TK]D-Fender | dan__t: a 3-way call started while being recorded would get all 3 |
| 04:29.28 | dan__t | How would I start it in the middle of a Record()? After Record() is called, and immediately after starting, does it automatically go to the next priority? |
| 04:30.01 | [TK]D-Fender | dan__t: actually, Record() does not record CALLS. it is a fixed recording for a solitary UNBRIDGED channel. |
| 04:31.43 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:31.57 | dan__t | Unbridged... so in this example, you couldn't use it against a channel that had three parties as part of it? |
| 04:32.07 | dan__t | you couldn't use it between two parties, either |
| 04:32.23 | [TK]D-Fender | dan__t: "core show application monitor" |
| 04:33.34 | *** join/#asterisk JimmyDee (n=jmdwyer@ppp-70-242-131-82.dsl.stlsmo.swbell.net) |
| 04:33.36 | dan__t | Oh bad-ass. |
| 04:33.39 | dan__t | That's perfect. |
| 04:35.07 | JimmyDee | question: is there a way to have an incoming caller select an extension and hook-flash 3 way call a cell phone? |
| 04:35.20 | JimmyDee | on pots w/3way |
| 04:36.10 | Kobaz | eh? |
| 04:36.46 | [TK]D-Fender | JimmyDee: What interface? |
| 04:37.04 | JimmyDee | well this is where I get lost |
| 04:37.26 | JimmyDee | I have done asterisk with sip, but the interested party has no interest in an internet bill |
| 04:37.43 | [TK]D-Fender | JimmyDee: You're talking hooks flash. that implies HARDWARE |
| 04:38.07 | JimmyDee | yes, and I honestly just need a point in a direction of what hardware that would be |
| 04:38.12 | [TK]D-Fender | JimmyDee: JimmyDee You seem to imply you're working with an analog LINE. So what piece of EQUIPMENT are you plugging it into? |
| 04:38.52 | JimmyDee | easy easy, I dont know, thats why I am asking the stupid questions |
| 04:39.11 | [TK]D-Fender | JimmyDee: thought : Digium TDM410P + FXO module |
| 04:39.13 | JimmyDee | yes analog line, 3way calling enabled |
| 04:39.15 | Kobaz | we need specifics! |
| 04:39.20 | Kobaz | [TK]D-Fender: :) |
| 04:40.11 | Kobaz | [TK]D-Fender: i think that call park just isn't meant to be used like this |
| 04:40.23 | [TK]D-Fender | JimmyDee: now before you go calling this a 3-way call, describe the call flow PRECISELY |
| 04:40.32 | Kobaz | [TK]D-Fender: i have a custom feature *1 that i want to put the other party into a parking lot, in position 1 |
| 04:40.54 | Kobaz | [TK]D-Fender: it runs a macro which only has two lines, one sets PARKINGEXTEN to 1, and the other does a Park() |
| 04:41.32 | Kobaz | [TK]D-Fender: the call gets parked, but then when the parking time expires, the phone calls itself |
| 04:42.05 | Kobaz | here's the log |
| 04:42.24 | Kobaz | ah |
| 04:42.28 | Kobaz | and i just crashed asterisk again |
| 04:42.40 | Kobaz | http://pastebin.com/m3b36bd63 |
| 04:42.50 | JimmyDee | ok that is exactly what I was looking for |
| 04:42.56 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 04:42.59 | Kobaz | so i repeatidly crash asterisk doing what i just did (in the paste) |
| 04:46.49 | flewid | anyone know why when i hit # to go to the directory, but CHANNEL(language)=fr is set, i still get the english prompt? (only for dir-intro.gsm) the rest all play in french |
| 04:47.00 | flewid | permissions are identical on all of them (asterisk:asterisk) |
| 04:48.55 | Jollyr01staup | [TK]D-Fender: Thank you very much for leading me in the right direction... |
| 04:49.40 | [TK]D-Fender | flewid: Show us "all of them" |
| 04:50.02 | [TK]D-Fender | Jollyr01staup: alrighty... |
| 04:50.04 | Jollyr01staup | [TK]D-Fender: My first mistake was assuming that genconf knows what it is doing... |
| 04:50.31 | [TK]D-Fender | Jollyr01staup: So you found a setting that matches? |
| 04:50.47 | [TK]D-Fender | Jollyr01staup: And no, I never trust conf geeeenerators like that. |
| 04:51.06 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 04:51.08 | Jollyr01staup | yes.. it ended upt being fxols |
| 04:51.16 | shmaltz | how do I block callerid using sip to an iax trunk? |
| 04:51.48 | shmaltz | i tried setcallerpres(prohib) and it doesn't work |
| 04:51.49 | shmaltz | Neither does setting cid(num) to 000 |
| 04:52.08 | [TK]D-Fender | shmaltz: maybe they don't LET you |
| 04:52.34 | shmaltz | ok then, I'm using teliax anyone know if they let? |
| 04:53.35 | [TK]D-Fender | shmaltz: Tried asked Teliax? I'm sure they'd know... |
| 04:53.51 | shmaltz | noone home at the moment :( |
| 04:54.49 | flewid | [TK]D-Fender. http://pastebin.ca/1307075 |
| 04:55.01 | flewid | notice it's using the 'fr' but always 'en' for dir-intro for some reason |
| 04:56.28 | Gopher_77 | I'm not having any success compiling the drivers necessary for my digium hardware for use with asterisk under netbsd. The hardware isn't being recognized. Can I get some help? |
| 04:57.59 | *** part/#asterisk JimmyDee (n=jmdwyer@ppp-70-242-131-82.dsl.stlsmo.swbell.net) |
| 04:58.51 | [TK]D-Fender | flewid: what line do I see this? |
| 04:59.14 | Gopher_77 | dmesg says "not configured" and pcictl says it's unrecognized |
| 04:59.30 | flewid | 149, 151, 138, 139, 154, 158 |
| 05:03.18 | [TK]D-Fender | flewid: well you're in a 3rd party AGI... I already don't trust it |
| 05:03.44 | flewid | hehe yeah, i'm just setting up a test directly with the regular directory to hunt down if this is the issue or if it's something else |
| 05:04.00 | flewid | i see a few other people on google having the same issue, but it was permissions, afaik mine are fine from what i pasted |
| 05:04.33 | [TK]D-Fender | flewid: Maybe it assumes a langue in its own config. Line the laguage of a user, etc... |
| 05:04.54 | [TK]D-Fender | flewid: I'm not going to debug this blind |
| 05:05.04 | flewid | well, it's kinda weird that only the intro plays wonky and the press 1 to hit the user works in french |
| 05:07.34 | [TK]D-Fender | flewid: untrusted AGI <- |
| 05:23.08 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 05:27.59 | *** join/#asterisk Wulfy814 (n=AskMe@17.wxfr23.xdsl.nauticom.net) |
| 05:36.45 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 05:37.01 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 05:41.34 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-da0ee4b728f9915d) |
| 05:47.22 | dan__t | [TK]D-Fender, found a problem where I need to specify a sounds directory... |
| 05:47.30 | dan__t | Trying to use Record(), and its looking for a 'beep'. |
| 05:49.11 | ricko73 | beep should be a standard sound |
| 05:49.11 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-242-212-37.phlapa.east.verizon.net) |
| 05:49.47 | dan__t | Yeah, but I'm having a problem with * finding sounds. |
| 05:50.02 | dan__t | I can play them if I specify an absolute path. |
| 05:50.20 | dan__t | I found 'astsounds', I'm going to play around with that a bit.... |
| 05:50.31 | ricko73 | usually, they are in /var/lib/asterisk/sounds where /var/lib/asterisk is defined in /etc/asterisk/asterisk.conf |
| 05:50.36 | dan__t | I know. |
| 05:50.42 | dan__t | Everything I've read says this, too. |
| 05:50.48 | [TK]D-Fender | dannot that the file folder layou for multi-lingual support HAS changed.... |
| 05:51.11 | [TK]D-Fender | dan__t: there have been 2 distinct dir structures to support this |
| 05:51.59 | dan__t | This I did not know. Can I get some sort of debug output other than core 10? Maybe something where it tells me the exact path its trying to look? |
| 05:52.27 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 05:52.59 | [TK]D-Fender | dan__t: read the docs for changes between 1.4 & 1.6 |
| 05:54.21 | dan__t | Nice, it says to look at 1.4's implementation. |
| 05:57.49 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 06:01.02 | dandate2 | so i finally got my * box working |
| 06:01.10 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:01.10 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:01.20 | dandate2 | if anyone wnats to chec kit out just call 1-201-620-6323 |
| 06:03.41 | [TK]D-Fender | dandate2: We believe you... |
| 06:03.50 | carrar | I need a TOLLFREE |
| 06:03.55 | carrar | Please set that up ASAP |
| 06:04.07 | [TK]D-Fender | s/believe/don't actually that that much about |
| 06:04.15 | [TK]D-Fender | :D |
| 06:05.00 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 06:05.05 | dan__t | heh. |
| 06:05.13 | dan__t | I'm about to start stabbing stuff. |
| 06:05.48 | dan__t | I can't debug to the point where I can find out exactly which absolute path this sound is trying to be called at? |
| 06:12.51 | ricko73 | wonders if dan__t has the 'astvarlibdir' variable set correctly in asterisk.conf |
| 06:12.58 | dan__t | That I do. |
| 06:13.07 | dan__t | I see my astdb being written to properly. |
| 06:13.39 | rdk5 | does anyone have any idea why I would get "SIP/2.0 404 Not Found" errors when trying to dial out? I am connected to voicepulse, but can't seem to dial out because I keep getting these errors... |
| 06:14.35 | [TK]D-Fender | rdkpastebin your failed call attemp with SIP debug enabled |
| 06:14.36 | [TK]D-Fender | ~pb |
| 06:14.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 06:15.40 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 06:16.28 | dan__t | Hrm. |
| 06:16.42 | ricko73 | dan__t: is languageprefix = yes set in asterisk.conf? |
| 06:17.07 | dan__t | According to the docs, the default is yes in 1.6 |
| 06:17.37 | dan__t | I've tried either way. |
| 06:18.39 | ricko73 | is this something you compiled from source or an installed package? |
| 06:20.06 | dan__t | Installed from a package. |
| 06:21.03 | rdk5 | [TK]D-Fender: www.pastebin.com/d30b4f0f9 |
| 06:21.28 | [TK]D-Fender | rdk5: WOW |
| 06:22.16 | rdk5 | [TK]D-Fender, wow? thats not what I like to hear :) |
| 06:23.47 | [TK]D-Fender | rdk5: EMPTY |
| 06:24.06 | rdk5 | oops, let me try again, i posted it on a diff computer |
| 06:25.44 | rdk5 | [TK]D-Fender, http://pastebin.com/d30b4f0f9 |
| 06:26.31 | rdk5 | that should work |
| 06:26.54 | [TK]D-Fender | rdk5: Looking for 18885551212 in from-internal (domain 10.0.0.5) <- fix your dialplan |
| 06:27.28 | rdk5 | i am using freepbx, i used the voicepulse module to make the dialplan |
| 06:27.42 | [TK]D-Fender | rdk5: FreePBX is NOT supported here |
| 06:28.03 | [TK]D-Fender | rdk5: If you need help using their interface they have their own channel #freepbx |
| 06:29.21 | rdk5 | [TK]D-Fender, hmm, ok, it is strange because I have an outgoing route headed to a trunk, voicepulse is registering the calls i believe... but i keep getting the 404 not found error |
| 06:30.01 | [TK]D-Fender | rdk5: No, * is telling you to get lost. |
| 06:30.20 | [TK]D-Fender | rdk5: Not VP |
| 06:30.44 | carrar | aha |
| 06:30.50 | carrar | BOOYAH |
| 06:31.22 | carrar | feels the love |
| 06:31.51 | carrar | rdk5, You should install Asterisk Source |
| 06:32.10 | rdk5 | carrar, I am working from asterisknow |
| 06:32.29 | carrar | ok, once you get Asterisk from Source intalled COME ON BACK!! |
| 06:32.47 | rdk5 | carrar, is something wrong with asterisknow? |
| 06:33.23 | carrar | depends what it's for |
| 06:33.44 | [TK]D-Fender | rdk5: Only your expectation of support for FreePBX here. |
| 06:33.44 | carrar | works for what it is |
| 06:33.44 | [TK]D-Fender | rdk5: Either way, you've got your answer |
| 06:34.16 | carrar | rdk5, personally I like Asterisk from Source as it gives you the most control over everything |
| 06:34.19 | rdk5 | carrar, it's for a very simple install. I've installed asterisk from source before, this was supposed to be a very simple server. |
| 06:34.36 | rdk5 | [TK]D-Fender, understandable, thanks for taking a look |
| 06:35.08 | carrar | You'll want to get your asterisknow support from #asterisknow |
| 06:35.25 | carrar | <PROTECTED> |
| 06:35.27 | [TK]D-Fender | carrar: #freepbx actually |
| 06:35.30 | carrar | oh |
| 06:35.33 | carrar | they renamed? |
| 06:35.41 | [TK]D-Fender | carrar: Given the distro channel kinda implies *-GUI |
| 06:35.54 | rdk5 | i will try, they just seem to be asleep |
| 06:35.55 | [TK]D-Fender | carrar: don't forget that *NOW includes BOTH GUI's now |
| 06:36.04 | carrar | I installed asterisk now once, then over wrote it |
| 06:36.11 | [TK]D-Fender | rdk5: that does not make this "level 2 support" |
| 06:36.36 | carrar | I'm just not a GUI person |
| 06:37.07 | MaliutaLap | I use a great GUI ... it's called vim |
| 06:37.09 | rdk5 | carrar, I'm normally not either, this was supposed to simplify a very simple setup.... oooops |
| 06:37.09 | carrar | cept when it comes to porn |
| 06:37.12 | carrar | j/k |
| 06:37.14 | carrar | not |
| 06:38.06 | carrar | rdk5, if you want simple easy to use gui, use switchvox! |
| 06:39.01 | [TK]D-Fender | carrar: If he can't handle FreePBX what makes any other GUI any easier? |
| 06:39.12 | carrar | they have phone support |
| 06:39.14 | carrar | heh |
| 06:39.18 | rdk5 | haha |
| 06:39.30 | carrar | excellent email support I might add too |
| 06:39.37 | [TK]D-Fender | carrar: ... you have a point. |
| 06:39.55 | [TK]D-Fender | carrar: A very sad point, but a point nonetheless |
| 06:39.59 | rdk5 | you guys are jumping to conclusions :) |
| 06:40.00 | carrar | heh |
| 06:40.15 | carrar | well sometimes switchvox is just the answer |
| 06:40.27 | [TK]D-Fender | rdk5: No, we're jumping on the BANDWAGON |
| 06:41.10 | rdk5 | last time i set up asterisk was about three years ago, and it was all non-gui, with lovely vim. It took awhile, but it worked. I was hoping that in 3 yrs, things had progressed to the point where something like asterisknow was usable to setup a small install in a short time. I think I was overoptimistic. |
| 06:41.48 | carrar | You see, it would be this mat that you would put on the floor and it would have different conclusions written on it that you could jump to. |
| 06:41.53 | dan__t | [pid 5436] stat("/usr/share/asterisk/sounds/en/beep.h264", 0x4016f070) = -1 ENOENT (No such file or directory) |
| 06:41.54 | dan__t | Very nice. |
| 06:43.07 | [TK]D-Fender | rdk5: FreePBX CAN be set up in a few odd minutes. You just seem to have failed to grasp how to set up outbound routes in it |
| 06:44.03 | [TK]D-Fender | rdk5: And there are dozens of sites out there that could show you hot to do this. |
| 06:44.08 | rdk5 | [TK]D-Fender, perhaps, somewhere. Although the VP module sets up outbound routes, and it did set them up. They just don't work. |
| 06:44.47 | carrar | Come back once you get it working with the #freepbx guys and lets us know what it was |
| 06:44.50 | [TK]D-Fender | rdk5: So not only using a GUI to set stuff up, but then a 3rd part module to set up the tool that was made so you don't have to set stuff up yourself... |
| 06:45.03 | [TK]D-Fender | rdk5: and THEN you wonder "where did it go wrong". |
| 06:45.16 | rdk5 | [TK]D-Fender, I was attempting to take my laziness to a new level :) |
| 06:45.27 | [TK]D-Fender | rdk5: Go take a look at any of the dozen web sites showing how to set that all up |
| 06:45.43 | [TK]D-Fender | rdk5: Putting the "suck" back into suckcess |
| 06:45.54 | [TK]D-Fender | rdk5: Get hopping little rabbit |
| 06:45.55 | carrar | haha |
| 06:46.09 | carrar | This channel is a such a release |
| 06:46.19 | dandate2 | can anyone tell me how do I configure x-lite to work with an inbound and outbound trunk? |
| 06:46.45 | rdk5 | [TK]D-Fender, carrar, do you guys feel better now? :) |
| 06:46.54 | carrar | dandate2, set it up just like you would a sip phone |
| 06:47.08 | carrar | rdk5, I'm smiling, so yeah |
| 06:47.22 | carrar | My wife thinks I'm nuts always laughing at the computer |
| 06:47.23 | dandate2 | but i have no option but to over write the existing host information for my DID provider with the outbound SIP provider |
| 06:47.40 | rdk5 | carrar, great, I am glad that I could improve your evening. Hopefully I was able to do the same for [TK]D-Fender. |
| 06:47.45 | carrar | xlite only allows 1 SIP Peer |
| 06:47.49 | carrar | unless you pay |
| 06:47.53 | [TK]D-Fender | rdk5: No. I have no particular sense of fulfillment from this. Go read the dozens of guides. You should already have your answer |
| 06:47.54 | dandate2 | oh |
| 06:48.00 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 06:48.06 | carrar | upgrade to BRIA |
| 06:48.12 | dandate2 | ok i'm going to look into paying |
| 06:48.14 | carrar | worth it |
| 06:48.22 | [TK]D-Fender | carrar: Wrong question |
| 06:48.27 | dandate2 | will i beable to get things like customer information, so if a customers calling i can know before hand |
| 06:48.28 | carrar | oh |
| 06:48.30 | dandate2 | or my rep |
| 06:48.47 | carrar | what was the question again |
| 06:48.48 | carrar | heh |
| 06:48.59 | [TK]D-Fender | carrar: You should be asking "Why the #&^$% are you setting up a softphone DIRECT to an ITSP. You're in friggen ASTERISK... * should be doing that job for you" |
| 06:49.04 | carrar | oh |
| 06:49.06 | carrar | haha |
| 06:49.07 | dandate2 | will the program transmit and store customer data so if someones a customer and calling the sales line the sales person knows before hand |
| 06:49.09 | [TK]D-Fender | carrar: BIG PRINT |
| 06:49.36 | carrar | dandate2, Why the #&^$% are you setting up a softphone DIRECT to an ITSP. You're n friggen ASTERISK... * should be doing that job for you!!!!!!!!!!! |
| 06:49.47 | rdk5 | [TK]D-Fender, do you really think I didn't do any googling before I asked a question in here? I followed the installation guide to a T, twice. Then spent some time googling my error messages... But thanks for taking a look. |
| 06:49.56 | dandate2 | i'm a little confused there, i'm using rapidvox.com for my outbound SIP and didforsale.com for my inbound |
| 06:50.15 | carrar | dandate2, upgrade to Bria |
| 06:50.24 | dandate2 | ok |
| 06:50.31 | carrar | then you can have more then 1 provider |
| 06:50.36 | dandate2 | i see |
| 06:50.47 | carrar | but yeah, setup your own asterisk box |
| 06:50.56 | dandate2 | do you know if that will have other cool features like notifying my rep of customer data before they pick up? |
| 06:50.57 | carrar | then just register to it and let it connect to all your providers |
| 06:51.08 | carrar | it's does a bunch of stuff |
| 06:51.10 | dandate2 | now i am very confused by that carrar |
| 06:51.12 | carrar | read their page |
| 06:51.27 | dandate2 | are you saying i can go without rapidvox for my outbound SIP and just let my * box handle it? |
| 06:51.33 | dandate2 | fender was saying that but i am way confused |
| 06:51.46 | *** join/#asterisk danielrm26 (n=daniel@24.96.188.216) |
| 06:51.53 | carrar | You have how many SIP Providers dandate2? |
| 06:51.58 | carrar | 2? |
| 06:52.03 | dandate2 | right one DID and out outbound |
| 06:52.13 | danielrm26 | Can you guys help with AsteriskNOW, or just Asterisk proper? |
| 06:52.20 | carrar | one DID? |
| 06:52.26 | carrar | same provider? |
| 06:52.31 | dandate2 | no different providers |
| 06:52.36 | [TK]D-Fender | danielrm26: GUI's are not supported here |
| 06:52.37 | carrar | heh |
| 06:52.39 | carrar | thats nutty |
| 06:52.39 | dandate2 | rapidvox for outbound and didforsale for inbound |
| 06:52.50 | [TK]D-Fender | carrar: far from |
| 06:53.16 | carrar | LNP/Port your number over to a SIP provider that can offer both in and out with your DID |
| 06:53.17 | danielrm26 | [TK]D-Fender: Cool, thanks. |
| 06:53.18 | dandate2 | fender are you saying i can do without rapid vox and somehow configure my PBX to do it? |
| 06:53.39 | [TK]D-Fender | dandate2: You setup * to register to your 2 ITSP's, take calls in and process from either, and accept calls from your softphone and send them out whatever resource is approriate. |
| 06:53.52 | [TK]D-Fender | dandate2: Your softphone does NOT talk to your providers directly <- |
| 06:54.02 | dandate2 | i see |
| 06:54.27 | dandate2 | but to have x-lite work both inbound and outbound over my 2 SIP trunks i'll need to upgrade no |
| 06:54.35 | carrar | no |
| 06:54.36 | [TK]D-Fender | dandate2: NO |
| 06:54.48 | [TK]D-Fender | dandate2: Your softphone doesn't know SHIT about your providers |
| 06:55.06 | carrar | (no if you use asterisk) |
| 06:55.14 | dandate2 | well i'm using asterisk |
| 06:55.18 | [TK]D-Fender | dandate2: when you dial a # * processes it and does what YOU tell it to |
| 06:55.32 | [TK]D-Fender | dandate2: For inbound, ASTERISK answers the call and does what you tell it to. |
| 06:55.34 | dandate2 | haha but i am such a noob at tellign it what to do, i mean i'm finally walking |
| 06:55.54 | carrar | You can run |
| 06:55.57 | [TK]D-Fender | dandate2: Go set up your SIP peers & registrations, and start coding your dialplan. Go read the book |
| 06:56.00 | [TK]D-Fender | ~book |
| 06:56.00 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 06:56.06 | [TK]D-Fender | dandate2: No shortcuts here. |
| 06:56.26 | [TK]D-Fender | dandate2: Chapter 5. master it |
| 06:56.30 | dandate2 | k |
| 06:56.49 | [TK]D-Fender | dandate2: The dialplan is THE most important part of *. |
| 06:57.27 | drmessano | He's jumping back and forth between here and #freepbx |
| 06:58.00 | [TK]D-Fender | drmessano: WHEE! |
| 06:58.06 | [TK]D-Fender | ~wglwat |
| 06:58.07 | jbot | somebody said wglwat was well, good luck with all that |
| 07:01.46 | drmessano | If he listened, he would know he cant follow whats in Chapter 5 |
| 07:06.56 | *** join/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 07:06.57 | dandate2 | err i had a hard timne understanding how chapter 5 can help me eliminate these issues.. |
| 07:07.09 | drmessano | Of course you did |
| 07:07.10 | dandate2 | it just reminded me of the extensions option in the freepbx gui |
| 07:09.03 | [TK]D-Fender | dandate2: Yes and what little you think you saw means nothing. You cannot compare call control when you di it yourself VS that of a GUI |
| 07:09.03 | dan__t | channel.c:3160 set_format: Unable to find a codec translation path from g729 to slin |
| 07:09.12 | dan__t | That's it. I had allowed=g729 in my sip.conf |
| 07:09.20 | dan__t | Don't want to be using that one anyway, huh. |
| 07:09.58 | dandate2 | i'm setting my on hold music in *, is it possible to have it play multiple mp3 files or do they all have to be tied into one? |
| 07:10.55 | [TK]D-Fender | danGuess what? X-lite doesn't SUPPORT G.729, and neither does * till you pay for the codec licenses for it |
| 07:11.14 | dandate2 | yeah but thats no laymans manual, i need to have hands on examples of what they are good for, i cannot comprehend the value of something looking at the formula alone |
| 07:11.23 | [TK]D-Fender | dandate2: And Yes * play froma folder full of sound files |
| 07:11.34 | dan__t | [TK]D-Fender, guess what? It shouldn't have been used as an example? |
| 07:11.34 | dan__t | heh |
| 07:11.45 | dandate2 | what is g729? |
| 07:11.50 | dan__t | BS. |
| 07:11.53 | dandate2 | ?? g729 |
| 07:11.57 | [TK]D-Fender | dan__t: bad autocomplete |
| 07:12.02 | dan__t | haha |
| 07:12.19 | dan__t | Yes, again, it didn't play from where it was *expected* to. |
| 07:12.26 | [TK]D-Fender | dan__t: But quite possibly applicable in full... for all I know |
| 07:12.31 | dandate2 | geez how do i know if i'm using g729? |
| 07:12.39 | dan__t | You don't specify it, then you won't miss it. |
| 07:12.54 | [TK]D-Fender | Bedtime... enough with the crazies here.... |
| 07:12.55 | dan__t | Learn from my 3 hour mistake. |
| 07:12.59 | dan__t | Later, thanks for the help. |
| 07:13.04 | dan__t | pbtththt. |
| 07:13.10 | dan__t | whatever, back to hacking. |
| 07:13.13 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-7b4490e46685535c) |
| 07:15.46 | *** join/#asterisk reneger (n=reneger@dslb-088-078-119-125.pools.arcor-ip.net) |
| 07:20.19 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:20.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:21.57 | dandate2 | whats the standard dial pattern ? |
| 07:21.57 | dandate2 | US |
| 07:22.16 | carrar | 10 digit dialing? |
| 07:22.22 | carrar | +1 |
| 07:22.32 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 07:22.38 | carrar | there are LOTS of patterns |
| 07:22.45 | SwK | in regex |
| 07:22.47 | dandate2 | right 1 area code |
| 07:22.58 | SwK | 1[2-9]\d{2}[2-9]\d{6} |
| 07:23.15 | carrar | like 011 |
| 07:23.23 | carrar | 01N |
| 07:23.32 | SwK | or [2-9]\d{2}[2-9]\d{6} or [2-9]\d{6} |
| 07:23.33 | carrar | 1010 |
| 07:23.38 | carrar | 0 |
| 07:23.46 | carrar | 00 |
| 07:23.47 | SwK | or 011\d+ |
| 07:24.01 | carrar | 911 etc.. |
| 07:24.04 | carrar | so many |
| 07:24.11 | SwK | not really |
| 07:24.14 | dandate2 | is 1NXXNXXXXXX |
| 07:24.14 | dandate2 | NXXNXXXXXX appropriate? |
| 07:24.19 | carrar | yeah |
| 07:24.26 | carrar | most common yes |
| 07:25.02 | SwK | dandate2, see nanpa.com for a definitive guide |
| 07:27.13 | dandate2 | i'm getting a different error now when trying to clal out, its telling me all circuits are busy |
| 07:27.26 | carrar | try again later then :) |
| 07:28.26 | carrar | could be your SIP peer is not setup correctly |
| 07:29.19 | dandate2 | sweet i got it owrking |
| 07:31.21 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 07:31.45 | drmessano | cool |
| 07:32.16 | dandate2 | ok my last step, configuring ATA device and plugging in plantronics amplifier with analog phone |
| 07:41.05 | dandate2 | could using massive wav files 15 minutes long cause sound problems? |
| 07:42.55 | dandate2 | is it better to use mp3? |
| 07:48.17 | Gopher_77 | dandate2: a 15 minute long .wav file could cause storage problems |
| 07:48.48 | Gopher_77 | dandate2: well, I suppose that depends on how much space you have on your drive |
| 07:49.39 | drmessano | How is a wav file gonna cause storage problems? |
| 07:50.26 | drmessano | Its one thing to say it's a larger file, but saying it like it's somehow going to set off a problematic reaction is just.. nonsensical |
| 07:50.37 | drmessano | Its going to be a big file. Period. |
| 07:51.19 | carrar | 1,024,000,000 bitrate |
| 07:51.30 | Gopher_77 | drmessano: Yes, running out of space would the the "storage problem", sorry |
| 07:51.46 | drmessano | Its either going to fit on the drive or its not |
| 07:52.05 | drmessano | and I am sure ONE WAV file isn't gonna bust a drive, unless this is 1991 |
| 07:52.28 | drmessano | 15min WAV at a sampling rate even usable by Asterisk will be under 100MB |
| 07:53.14 | Gopher_77 | mp3 though is a patented format and would probably be decoded by a for-pay decoder |
| 07:53.15 | drmessano | If thats gonna create a storage problem, I would suggest "panic" |
| 07:53.21 | Gopher_77 | better to find a free format |
| 07:53.40 | drmessano | That has nothing to do with it |
| 07:53.41 | carrar | use format from asterisk-addons for mp3 |
| 07:53.53 | drmessano | The issue would be CPU usage |
| 07:54.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:54.18 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:54.30 | Gopher_77 | ogg is comparable to mp3 |
| 07:54.37 | drmessano | You almost forgot "It a file format that has a W in it", since you've hit all the nonusable arguments |
| 07:54.44 | Gopher_77 | drmessano: yes, you just wanted to ramble on about the same thing |
| 07:54.49 | drmessano | Yes, and still requires CPU overhead |
| 07:54.58 | drmessano | No, my comments actually make sense |
| 07:54.58 | Gopher_77 | drmessano: yes |
| 07:55.30 | Gopher_77 | drmessano: are you normally this ignorant? |
| 07:55.44 | drmessano | Do you normally make such stupid arguments? |
| 07:56.00 | drmessano | Like having "storage problems" |
| 07:56.01 | Gopher_77 | drmessano: seems like you're the only one arguing here |
| 07:56.01 | carrar | Gopher_77, your statement was clearly wrong about mp3 with regards to asterisk |
| 07:56.18 | Gopher_77 | carrar: wrong how? |
| 07:56.25 | drmessano | I think there's a new post on Digg, Gopher_77.. Your people are calling |
| 07:56.28 | carrar | <Gopher_77> mp3 though is a patented format and would probably be decoded by a for-pay decoder |
| 07:56.40 | Gopher_77 | carrar: is it not? |
| 07:56.53 | carrar | no need for a comerical decode when using mp3's with asterisk |
| 07:56.58 | carrar | decoder |
| 07:57.07 | drmessano | DEcoder |
| 07:57.11 | drmessano | not ENcoder |
| 07:57.22 | Gopher_77 | carrar: oh, asterisk decodes it itself then? |
| 07:57.25 | carrar | the subject was playing a file |
| 07:57.31 | drmessano | I would suggest googling for "Decoding MP3" and "ignorance" |
| 07:57.53 | Gopher_77 | yeah, how about drmessano and ignorance |
| 07:57.58 | Gopher_77 | I'm trying to learn here |
| 07:58.03 | drmessano | carrar: I thought it was using a RAID for a WAV file |
| 07:58.13 | carrar | Check out format in the asterisk-addons |
| 07:58.30 | carrar | s/format/format_mp3/ |
| 08:00.12 | drmessano | drmessano and ignorance is up to 89 hits on Google |
| 08:00.21 | drmessano | When I get to 100, I get my Blue Star |
| 08:00.24 | drmessano | YAY! |
| 08:00.36 | Gopher_77 | at least 90 now |
| 08:00.43 | carrar | If you have the choice use whatever format your call is in |
| 08:00.54 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 08:01.04 | carrar | less overehad on * |
| 08:01.06 | carrar | head |
| 08:01.13 | drmessano | Yep |
| 08:01.19 | carrar | or just use WAV |
| 08:01.23 | carrar | sounds better |
| 08:01.25 | drmessano | No |
| 08:01.31 | drmessano | "storage problems" dude |
| 08:01.34 | carrar | haha |
| 08:01.36 | carrar | oh yeah |
| 08:01.38 | Gopher_77 | lol |
| 08:01.46 | carrar | for those RLL 40 meg drives |
| 08:01.53 | drmessano | Copy a WAV over.. RAID failure, I guess |
| 08:02.18 | Gopher_77 | mostly a waste of space normally |
| 08:02.29 | drmessano | Not for the tradeoff |
| 08:02.41 | drmessano | WAV is a much better sounding format, and less CPU to play it |
| 08:02.44 | drmessano | Storage is cheap |
| 08:02.48 | Gopher_77 | unless you're really that concerned about CPU |
| 08:02.50 | dandate2 | why is it files that i upload to music on hold will not show up? |
| 08:03.23 | drmessano | Get 5 or 6 callers on hold in a queue and lets talk about CPU |
| 08:03.24 | Gopher_77 | hey, maybe he has a 486 with a 40MB hard disk |
| 08:03.27 | drmessano | MP3 is horrible |
| 08:03.33 | drmessano | OGG isn't much better |
| 08:03.40 | Gopher_77 | they're about the same |
| 08:03.53 | drmessano | Depends on the decoder and application |
| 08:03.54 | dandate2 | wait messano i am going to have 5-6 callers on hold in a queue, in that case should i just use mp3? |
| 08:04.02 | drmessano | WTF |
| 08:04.07 | Gopher_77 | of course what bitrate do you use, and do you hear it over a phone line? |
| 08:04.09 | drmessano | WAV <-- CAN U SPEAK ENGLIS? |
| 08:04.21 | drmessano | WAV = LESS CPU |
| 08:04.25 | dandate2 | oh ok |
| 08:04.26 | drmessano | MP3 = MORE CPU |
| 08:04.32 | dandate2 | i see |
| 08:04.37 | drmessano | Fucking read when you ask questions and people answer them |
| 08:04.49 | dandate2 | yes sir |
| 08:04.55 | drmessano | Dont patronize me |
| 08:05.07 | Gopher_77 | I suppose if you have several people on, and you're constantly playing the music, yes, drmessano has a good point |
| 08:05.43 | dandate2 | so in the on hold messaging tab i created a music category called beethoven, i tried to upload 3 wav files of his symphony but they won't show up. howevewr in the default category there shows 3 files that came with * |
| 08:06.18 | drmessano | Thats a FreePBX question and has nothing to do with Asterisk |
| 08:06.35 | M-33 | hello, how can i enable three-way calling and how to activate it over the softphone? |
| 08:06.42 | carrar | dandate2, wrong channel |
| 08:07.05 | drmessano | and so he pastes in there |
| 08:07.07 | carrar | come back when you install Asterisk from Source |
| 08:07.31 | carrar | time to put him on ignore |
| 08:08.44 | carrar | M-33, read the manual on the phone |
| 08:09.20 | M-33 | carrar, so you mean its already enabled by default on asterisk? |
| 08:09.24 | carrar | yes |
| 08:09.29 | M-33 | cool thanks |
| 08:11.23 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 08:18.06 | *** join/#asterisk bmg505 (n=leon@196-209-77-132-tbnb-esr-2.dynamic.isadsl.co.za) |
| 08:31.14 | *** join/#asterisk unasi7 (n=unasi7@84-75-23-200.dclient.hispeed.ch) |
| 08:34.15 | *** join/#asterisk Rico29 (n=rico@sto93-5-88-174-207-65.fbx.proxad.net) |
| 08:34.17 | Rico29 | hi |
| 08:34.25 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:39.53 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 08:42.13 | *** join/#asterisk matt_ (n=matt@mattspc.ipv6.mattstone.net) |
| 08:50.02 | *** join/#asterisk ibm2 (n=Administ@196.203.44.3) |
| 08:54.02 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:55.05 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.119.232) |
| 08:55.22 | DelphiWorld | hi my friends |
| 08:55.34 | DelphiWorld | please any asterisk developers conference to check it Out ? |
| 08:55.59 | *** part/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com) |
| 08:57.03 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 08:57.33 | DelphiWorld | brian: hi |
| 09:01.09 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 09:02.04 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 09:02.57 | DelphiWorld | my friends: i'm blind. please anyone here have the pocibility to start with me about asterisk ? |
| 09:03.44 | khronos | Hi. |
| 09:04.04 | khronos | You'll need acces to a linux machine. |
| 09:05.04 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 09:05.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:06.16 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 09:08.08 | *** join/#asterisk DarkRift (n=dark@65.92.167.151) |
| 09:09.37 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
| 09:19.48 | *** join/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 09:20.02 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 09:20.09 | HeMan | Hi! I can't get Background to work in a macro |
| 09:21.05 | HeMan | I've tried Background(demo-instruct|macro-mymacro) |
| 09:21.33 | HeMan | and an exten => i,1,SayDigits(1) |
| 09:22.08 | HeMan | and exten => 1,1,SayDigits(1) |
| 09:23.50 | HeMan | I've also tried Background(demo-instruct|mymacro) |
| 09:24.17 | HeMan | I get "Invalid extension '1', but no rule 'i' in context 'mycontext'" |
| 09:25.27 | HeMan | Ah, found it! |
| 09:26.12 | HeMan | should be Background(demo-instruct|||macro-mymacro) |
| 09:26.39 | HeMan | the context must be the 4:th argument |
| 09:33.07 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 09:33.07 | *** join/#asterisk bagpuss_thecat (n=bagpuss_@lodge.glasgownet.com) [NETSPLIT VICTIM] |
| 09:38.34 | *** join/#asterisk thansen (n=thansen@h-68-164-180-2.atlngahp.covad.net) |
| 09:40.11 | *** join/#asterisk dan__t (n=dant@ip70-162-189-250.ph.ph.cox.net) |
| 09:40.13 | dan__t | Hi. |
| 09:44.20 | *** join/#asterisk xrmx__ (n=rm@host220-252-dynamic.9-87-r.retail.telecomitalia.it) |
| 09:51.51 | *** join/#asterisk BipBip (n=BipBip@static-b5-253-32.telepac.pt) |
| 09:54.26 | *** join/#asterisk JerJer (n=PhatJ@24-236-207-64.dhcp.aldl.mi.charter.com) |
| 10:04.06 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 10:06.42 | *** join/#asterisk IPGHOST (n=IPGHOST@116.71.170.25) |
| 10:08.15 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:09.11 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 10:10.34 | beniwtv | Hi all... Is it possible to change the SQL queries for the asterisk mysql realtime driver? (I don't want to write data to my MySQL slaves) If not, is there a way to tell Asterisk to use an internal DB to store IP addresses and ports? |
| 10:10.59 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 10:18.45 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 10:25.05 | dan__t | erm, I haven't read anything on that. Are you simply trying to get data to go to one or more MySQL servers? |
| 10:25.07 | dan__t | er, two or more. |
| 10:28.07 | beniwtv | dan_t: When a SIP user authenticates via realtime to *, it updates the DB with the IP address, port and regseconds. However, the DB I'm using is a slave server, so it can't use UPDATE queries. |
| 10:28.45 | dan__t | Ah hah. Sorry, still new to *, especially realtime. |
| 10:29.42 | dan__t | With similar projects housing similar information in a similar situation, I've had success with using multi-master replication. I don't know if that might help you in this particular situation, but hey, who knows... |
| 10:29.56 | dan__t | That almost rhymes. I'm ill, I need to go to sleep :( |
| 10:31.25 | beniwtv | dan__t: Yeah, however, this replication setup we hav is years old, so I don't want to touch it, if possible. |
| 10:32.42 | *** join/#asterisk dandate2 (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net) |
| 10:33.00 | dandate2 | w00t i have configured MOH for free-pbx!! i am a real nerd now |
| 10:33.12 | beniwtv | dan__t: I found an option, rtupdate=yes ("Send registry updates to database using realtime? (yes|no) If set to yes, when a SIP UA registers successfully, the ip address, the origination port, the registration period, and the username of the UA will be set to database via realtime. ") |
| 10:33.15 | beniwtv | Could that be it? |
| 10:33.43 | beniwtv | However, I'm not sure if then * stores the IP in an internal DB or something |
| 10:34.53 | dan__t | That I cannot answer, I'm sorry. |
| 10:34.55 | dan__t | Give me a week :) |
| 10:35.05 | dan__t | I need to get a little nap in here, good luck. |
| 10:35.52 | beniwtv | dan__t: Don't worry :) |
| 10:37.16 | JerJer | beniwtv: do a master-master replication |
| 10:38.00 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 10:38.49 | beniwtv | JerJer: As already said, that's sadly not an option. |
| 10:40.20 | JerJer | good luck then |
| 10:40.38 | beniwtv | Now, I've tried a call via a SIP phone with rtupdate=no, and it seemed to work fine. But I'd still like to know what implications turning off this function brings. Would * still be able to find the soft phone if called? What about NAT? |
| 10:42.04 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:44.15 | *** join/#asterisk dec_ (n=tom@unaffiliated/dec) |
| 10:45.03 | dec_ | Can anyone help diagnose why our licensed (from digium) g729 codec 'just stopped working' after a server reboot? |
| 10:45.42 | beniwtv | dec__: Anything in the logs? |
| 10:46.33 | dec_ | the actual error we're getting is "channel.c: Unable to find a codec translation path from g729 to gsm", but I get a "codec_g729a.c: Failed to initialize G.729 copy protection!" when asterisk loads the codec module |
| 10:47.38 | dec_ | absolutely nothing has changed on this system (yes, I'm sure) |
| 10:47.53 | beniwtv | dec_: Hmm... that seems to be a problem with your license. Did you change your Ethernet NIC's, MAC addresses or something? |
| 10:47.55 | dec_ | the codec module is still there, the license is still in /var/lib/asterisk/licenses/ |
| 10:48.06 | dec_ | nope, same NICs with same MAC |
| 10:48.51 | beniwtv | Did you do any system update? |
| 10:49.07 | dec | None at all. |
| 10:49.30 | dec | (I checked the logs :)) |
| 10:50.09 | beniwtv | Maybe you should contact Digium in that case, they may be able to help you |
| 10:50.27 | dec | OK, that was my next hoep |
| 10:50.29 | dec | hope* |
| 10:59.13 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 11:03.18 | dec | Email sent to digium support, hopefully they are prompt... |
| 11:04.12 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 11:06.14 | ruben23 | hi have error compiling zaptel http://pastebin.com/m74dae085 |
| 11:08.00 | ruben23 | anyone have idea on this |
| 11:12.38 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:24.07 | *** join/#asterisk Faustov (i=fst@gentoo/user/faustov) |
| 11:25.10 | Faustov | hello, what would be the recommended way to restrict certain calls at given hours? Does it have to be scripted with conditionals? |
| 11:25.20 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:28.07 | Faustov | i guess it can be easily done with cron and shell scripting swapping extensions.conf, however i have a feeling there must be a nicer way... |
| 11:29.43 | *** join/#asterisk qdk_ (n=qdk@78.156.217.143.bredband.3.dk) |
| 11:31.50 | JerJer | Faustov: GotoIfTime ? |
| 11:33.08 | yang | ~GotoIfTime |
| 11:33.55 | yang | ~GotoIfTime is at http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime |
| 11:33.56 | jbot | okay, yang |
| 11:34.51 | Faustov | checking |
| 11:35.41 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 11:35.43 | tengulre | hi,all |
| 11:36.57 | JerJer | moo |
| 11:38.37 | Faustov | seems exactly what i needed, i wonder why i couldn't google it out |
| 11:38.44 | Faustov | thanks JerJer |
| 11:39.00 | JerJer | no problemo |
| 11:39.01 | Faustov | funny with the easter predictions :> |
| 11:57.56 | *** join/#asterisk chendy (n=chendy@121.34.152.100) |
| 12:12.42 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.70.151) |
| 12:13.08 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 12:16.05 | mandh | Hi , can i make different ring tone on the IP phnes for each group on incomming calls from external telphone lines |
| 12:17.07 | JerJer | depends on the ip phone |
| 12:17.36 | mandh | what that fetures named ? |
| 12:18.25 | mandh | features |
| 12:20.22 | anonymouz666 | distinctive ring? |
| 12:21.29 | anonymouz666 | if your phone does not support it, you can play an announcement when someone pickup the phone, but I agree that distinctive ring is better. |
| 12:22.17 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 12:22.34 | mandh | anonymouz666, so i must first see that the phone support the distinctive |
| 12:22.47 | mandh | it is Thomson st2030 |
| 12:25.13 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) |
| 12:46.49 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 12:47.37 | ScriptFanix | did anone have a catchall (eg. for invalid numbers) working with a queue ? |
| 12:49.15 | ScriptFanix | I got a queue which rings 3 extensions, and if I use some kind of catchall extension (at the end of the dialplan), the queue sends me to the catchall extension |
| 12:50.58 | *** join/#asterisk ManxPower (n=manxpowe@234.sub-70-214-180.myvzw.com) |
| 12:53.19 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 12:54.15 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 12:54.16 | *** join/#asterisk sasargen (n=chatzill@174-152-61-130.pools.spcsdns.net) |
| 12:54.21 | IsUp | hiya |
| 12:54.41 | IsUp | anyone knows how can i create .h263 files for vm prompts? |
| 12:55.01 | ScriptFanix | (dialplan looks like this: http://paste.quarantedeux.net/196 |
| 12:57.13 | ScriptFanix | (queue members are defined as Local/100@appel-sortant, Local/101@appel-sortant, Local/102@appel-sortant) |
| 12:58.22 | ScriptFanix | IsUp: using the Record() function in your dialplan ? |
| 12:59.42 | ScriptFanix | IsUp: http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 13:02.57 | IsUp | ScriptFanix: record already works, but i want to create custom prompts. i have video files and want to convert them to h263 |
| 13:09.42 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 13:14.04 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:14.15 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:18.19 | *** join/#asterisk tokozedg (n=h3kt0r0@85.118.98.122) |
| 13:18.52 | tokozedg | hi all, everyone knows online player to play asterisk records(gsm ) files ? |
| 13:23.31 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:24.22 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 13:25.35 | phix | hey |
| 13:27.24 | *** join/#asterisk janinges (n=janinge@58.85-200-255.bkkb.no) |
| 13:28.15 | *** join/#asterisk nexu (n=nexu@a82-95-43-129.adsl.xs4all.nl) |
| 13:28.30 | eppigy | hello |
| 13:28.32 | eppigy | i am dave |
| 13:30.12 | janinges | hi dave |
| 13:30.43 | eppigy | hello |
| 13:31.28 | *** join/#asterisk janinge (n=janinges@kramer.zf.no) |
| 13:40.27 | *** join/#asterisk docelmo (n=chatzill@206-248-239-194.unassigned.ntelos.net) |
| 13:44.08 | mark_csi | hi all, anyone know how to apply a digium patch to an existing asterisk server? |
| 13:50.13 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:50.13 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:52.24 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:53.12 | [TK]D-Fender | mark_csi: like? |
| 14:01.25 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:01.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:08.28 | tzafrir_laptop | PostgreSQL seems to have all the right connections in Australia: http://lwn.net/Articles/314724/ ;-) |
| 14:10.35 | yang | Heh, I discovered one VOIP uplink which produces an error - Comfort noise support incomplete in Asterisk (RFC 3389) |
| 14:11.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:15.42 | *** join/#asterisk anonymouz666 (n=anonymou@189.36.179.210) |
| 14:18.59 | Kobaz | man, i kepe breaking asterisk |
| 14:20.59 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:21.02 | mark_csi | [TK]D-Fender: I've found a specific patch that sorts out a problem I have, I've downloaded the file but don't know what to do with it. |
| 14:23.20 | [TK]D-Fender | mark_csi: Some details would be nice... |
| 14:24.04 | *** part/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 14:25.15 | Kobaz | how do i reopen as asterisk bug on the tracker |
| 14:25.22 | Kobaz | i dont see any way to add a comment or reopen a bug |
| 14:25.29 | Kobaz | or should i just submit a new one? |
| 14:25.50 | Kobaz | http://pastebin.com/m1abd268b |
| 14:26.09 | *** join/#asterisk thansen (n=thansen@firewall.ssisecurity.com) |
| 14:28.00 | *** join/#asterisk dec_ (n=tom@unaffiliated/dec) |
| 14:28.10 | Kobaz | anyone? |
| 14:28.54 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:29.14 | Kobaz | it's very similar (if not a regression) of this bug: http://bugs.digium.com/bug_view_page.php?bug_id=12359 |
| 14:30.59 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:31.45 | mark_csi | [TK]D-Fender: sorry, this is the bug relating to my issue: http://bugs.digium.com/view.php?id=9264 |
| 14:34.17 | [TK]D-Fender | mark_csi: this was fixed a good while ago. Just upgrade |
| 14:34.40 | Kobaz | do de do |
| 14:35.07 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:35.17 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 14:35.25 | tzafrir_laptop | mark_csi, what version of Zaptel do you have? |
| 14:35.37 | Kobaz | i guess i'll just submit a new bug? |
| 14:36.14 | tzafrir_laptop | Kobaz, what issue? |
| 14:36.16 | [TK]D-Fender | Kobaz: What version are you running? |
| 14:36.23 | tzafrir_laptop | ah, ok |
| 14:36.27 | Kobaz | 1.4.22 |
| 14:36.32 | Kobaz | http://pastebin.com/m1abd268b |
| 14:37.29 | *** join/#asterisk xrmx__ (n=rm@host220-252-dynamic.9-87-r.retail.telecomitalia.it) |
| 14:39.48 | Kobaz | it's probably similar to the agi crash i discovered last night |
| 14:40.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:43.10 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:44.39 | Kobaz | it's actually two problems |
| 14:44.57 | Kobaz | == Parked SIP/5506-081de6f0 on 1@parkedcalls. Will timeout back to extension [basic] s, 1 in 10 seconds |
| 14:45.23 | Kobaz | the timeout extension is wrong |
| 14:50.12 | mark_csi | [TK]D-Fender: I'm running 1.6.0.1 |
| 14:51.02 | mark_csi | tzafrir_laptop: I'm running dahdi, just trying to get version now. |
| 14:51.19 | *** join/#asterisk mog (n=mog@nat/digium/x-e1d9fbf86dde0c0b) |
| 14:51.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:51.20 | Kobaz | i have a feeling park just wasn't meant to be used this way |
| 14:51.25 | Kobaz | but it shouldn't crash nonetheless |
| 14:51.42 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com) |
| 14:51.53 | tzafrir_laptop | mark_csi, this has been fixed before dahdi got released |
| 14:53.37 | mark_csi | tzafrir_laptop: pants, just can't get round this 'CALLERID timed out' error, just hangs the calls up |
| 14:54.17 | Kobaz | ack, found another crash |
| 14:54.18 | mark_csi | tzafrir_laptop: I've spoken to digium about it but they've no idea, also replaced the analogue card. |
| 14:54.27 | Kobaz | this one looks like memory corruption |
| 14:55.00 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 14:55.05 | Kobaz | well, you guys are busy, so i'll just submit new bug reports... i don't see any way at all to reopen an old bug |
| 14:55.57 | *** join/#asterisk stintel (i=stijn@madwifi/support/stintel) |
| 14:56.20 | stintel | moin all |
| 14:56.23 | stintel | quick question |
| 14:56.33 | stintel | -- Channel 0/1, span 1 got hangup request, cause 50 |
| 14:56.45 | eppigy | ISDN CAUSE CODES |
| 14:56.46 | stintel | where would one find the explanation for that cause 50 ? |
| 14:56.48 | eppigy | GOOGLE.COM |
| 14:56.49 | ricko73 | moin moin to you too |
| 14:56.51 | stintel | eppigy: thx |
| 14:56.54 | eppigy | np |
| 14:56.58 | stintel | eppigy: no need to shout :P |
| 14:57.01 | eppigy | sorry |
| 14:57.06 | eppigy | i type in all caps all day |
| 14:57.09 | eppigy | CRUISE CONTROL |
| 14:57.12 | stintel | lol |
| 14:57.20 | eppigy | 8[] |
| 14:57.48 | ricko73 | eppigy that would be fine if you talked in acronyms |
| 14:58.10 | eppigy | S.L.O.S.S.I.N |
| 14:58.16 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:58.42 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-2ba79f0e5ac7b9d3) |
| 15:00.13 | Kobaz | allrightey, new bug http://bugs.digium.com/view.php?id=14228 |
| 15:01.51 | stintel | so another question ... anybody any idea why I get isdn cause code 50 when dialing a number that actually is subscribed? dialing other numbers with same format works fine, and dialing that number that gives cause 50 on a "normal" phone works fine too |
| 15:02.12 | stintel | hmm |
| 15:02.35 | stintel | mixing up parts of sentences again it seems |
| 15:04.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:06.50 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:07.51 | jameswf | ~weather KPHX |
| 15:08.15 | jameswf | just wanted to brag a little :) |
| 15:09.07 | jameswf | high today of like 74 ... |
| 15:11.06 | Kobaz | man that's way too hot |
| 15:11.21 | Kobaz | it's supposed to be a scorcher today at 15F |
| 15:11.32 | ricko73 | jameswf: stfu |
| 15:11.48 | ricko73 | High of 11F and dropping |
| 15:12.04 | jameswf | MN suppose to be high -3 today |
| 15:12.28 | Kobaz | ~weather KALB |
| 15:12.53 | Kobaz | 27?.. hmm |
| 15:12.56 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:13.04 | Kobaz | my weather dockapp must have old data, it says 12F with a high of 15 |
| 15:13.09 | jameswf | -2.8 + 30mph winds = -12? |
| 15:14.04 | Kobaz | ~weather KSCH |
| 15:14.30 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733320.dsl.bell.ca) |
| 15:14.34 | Kobaz | oh okay, that is right... wow it's 10 degrees warmer 30 min south |
| 15:15.07 | ricko73 | ~weather KSBM |
| 15:15.39 | ricko73 | at least it's sunny ;) |
| 15:15.43 | Kobaz | haha |
| 15:15.56 | anonymouz666 | good luck for you guys it's more than +30 C here. |
| 15:16.04 | stintel | ~weather EBBR |
| 15:16.22 | Kobaz | the coldest town in all of new york |
| 15:16.31 | Kobaz | ~weather KSLK |
| 15:16.45 | Kobaz | is warmer? that makes no sense |
| 15:17.28 | Kobaz | oh well |
| 15:17.32 | Kobaz | back to breaking asterisk |
| 15:17.39 | NoxIn- | ~weather LFMN |
| 15:22.03 | *** join/#asterisk lclimber (n=lcanelon@212.183.204.76.static.user.ono.com) |
| 15:23.05 | lclimber | hello everyone, i have a question, is there a way to connect a movil device (cell phone) to an asterisk server? |
| 15:25.05 | Kobaz | gsm gateway |
| 15:28.24 | NovceGuru | jameswf: where you at in phx? |
| 15:29.54 | jameswf | Tempe |
| 15:30.04 | NovceGuru | nice |
| 15:30.50 | *** join/#asterisk monstro (n=monstro@201-68-37-185.dsl.telesp.net.br) |
| 15:30.55 | monstro | Hi folks, |
| 15:31.07 | monstro | Im don't can run asterisk! |
| 15:31.21 | monstro | it display the message: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk/asterisk.ctl exist?)" |
| 15:31.32 | [TK]D-Fender | monstro: How did you start it? |
| 15:31.35 | NovceGuru | I would like to spend more time in phoenix atm, esp with this weather |
| 15:31.57 | monstro | [TK]D-Fender, with command: asterisk -r -x reload |
| 15:32.08 | monstro | Im need reload it |
| 15:32.24 | [TK]D-Fender | monstro: that doesn't START *. That tries to connect to a RUNNING * daemon |
| 15:32.40 | [TK]D-Fender | monstro: how did you START *, and what user are you logged in as trying to connect to it now? |
| 15:32.56 | NovceGuru | jameswf: how close are you to 4th st and mcdowell :) |
| 15:33.40 | *** join/#asterisk Caede (n=caede@204.94.248.216) |
| 15:35.12 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-757a7d80f3298645) |
| 15:35.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:35.17 | *** join/#asterisk Sargun_Screen (n=sargun@208.106.98.2) |
| 15:35.28 | Sargun_Screen | Anyone know what's going on with: http://www.digium.com/en/mediacenter/viewpress/Digium-and-Skype-Collaborate-to-Bring-Skype-to-Business-Phone-Systems |
| 15:37.26 | [TK]D-Fender | Sargun_Screen: Still in Beta |
| 15:38.13 | Sargun_Screen | how long in beta |
| 15:38.30 | ricko73 | Sargun_Screen: listen to the VUC call from a few weeks ago. Jtodd discussed as much as he could publically |
| 15:38.38 | coppice | until the marketing dept says its ready |
| 15:39.06 | ricko73 | http://www.voipusersconference.org |
| 15:40.08 | ricko73 | I apologize, it was Steve Sokol who was on the call |
| 15:40.22 | ricko73 | http://feeds.feedburner.com/~r/AstUser/~3/495745435/TS-170693.mp3 |
| 15:41.00 | rickross | I installed 1.6.0.3 last night, and curiously it seems to be working the host machine harder than 1.4.x did - is there any easy way yo discover why * is increasing the load average on the host system? |
| 15:41.03 | *** join/#asterisk chendy (n=chendy@121.34.152.100) |
| 15:41.36 | Caede | Probably a stupid question... why does the accountcode field in the CDR module always get truncated to 20 chars? Is there a way to adjust this? I can find a constant anywhere. |
| 15:42.35 | jameswf | 20-30min |
| 15:43.28 | codefreeze-lap | Caede: The field lengths are set in include/asterisk/cdr.h; I think that would be your bottleneck. The DB tables need also to be defined to hold longer strings. |
| 15:44.04 | Caede | Yeah, DB table was already updated. I'll check there -- thanks. Always the most obvious place. |
| 15:45.19 | Caede | Awesome. Lifesaving, thanks. |
| 15:46.05 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-5935f1988aee3055) |
| 15:46.49 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 15:47.21 | *** join/#asterisk grndslm (n=grndslm@24-119-80-142.cpe.cableone.net) |
| 15:47.36 | stencil | ~itsp |
| 15:47.37 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 15:51.06 | *** join/#asterisk sasargen (n=chatzill@174-152-61-130.pools.spcsdns.net) |
| 15:53.10 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
| 15:54.05 | rgsteele||work | I'm migrating to a setup where I have a static endpoint, and have traditionally always had dynamic endpoints. Do I still need a 'register => ...' line in my sip.conf? Or is there another way to just specify the IP of the gatekeeper on the other end? |
| 15:59.11 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 16:00.19 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a970440124c1c189) |
| 16:00.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:00.37 | [TK]D-Fender | rgsteele||work: You do not need to register if you can tell your provider a fixed IP / host. |
| 16:01.26 | rgsteele||work | Yeah, but I need to specify the endpoint on the telco side somehow in sip.conf, right? |
| 16:02.03 | rgsteele||work | I guess I'm not sure what needs to change in the sip.conf, other than the register line being unnecessary. |
| 16:05.04 | [TK]D-Fender | rgsteele||work: huh? |
| 16:05.27 | [TK]D-Fender | rgsteele||work: Registering is to tell THEM where we are. If they already know, then you don't need a register line |
| 16:05.40 | [TK]D-Fender | rgsteele||work: You still need a peer entry to auth calls in & out like normal. |
| 16:07.03 | rgsteele||work | Ah, okay. That answers my question. Nothing else is really needed then except a peer definition of the SIP peer. |
| 16:07.16 | rgsteele||work | Thanks. |
| 16:07.32 | *** join/#asterisk Sargun (n=Sargun@66.151.148.225) |
| 16:09.51 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:11.41 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:12.32 | putnopvut | what prompted jbot to say that? |
| 16:12.52 | bmoraca | when using a T1 PRI, is it possible to number the B channels from 0 to 22 instead of 1 to 23? |
| 16:13.58 | eppigy | D: |
| 16:14.30 | stintel | putnopvut: he's developing his own will probably ツ |
| 16:14.34 | stintel | AI ftw :P |
| 16:15.27 | putnopvut | heh |
| 16:15.47 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 16:15.55 | stencil | hello guys, the jbot's ~itsplist-ca has been the same for at last twelve months are there any new good ITSPs in Canada? |
| 16:17.06 | SQLDarkly | Hey. I noticed there is a stated issue for Dahdi 2.x not compiling and exitign with error code 2. I have found no work around for this. Has anyone successfully compiled dahdi-linux-complete-2.1.0.2 or .3? |
| 16:17.13 | SQLDarkly | If so on what kernel? |
| 16:17.41 | stencil | putnopvut: I private messaged jbot that is what probably triggered that outburst |
| 16:18.25 | [TK]D-Fender | bmoraca: Yes |
| 16:18.36 | [TK]D-Fender | bmoraca: Sorry, I misread that. No. |
| 16:18.57 | [TK]D-Fender | bmoraca: Zaptel/DAHDI's numbering scheme starts at "1" |
| 16:19.58 | SQLDarkly | FYI my kernel is verions -2.6.16.60-0.31-bigsmp |
| 16:24.12 | tzafrir_laptop | what distro is that? |
| 16:24.22 | bmoraca | so there's no possible way to start it at 0? that's unfortunate. Oh well. |
| 16:24.38 | tzafrir_laptop | bmoraca, why do you need it to? |
| 16:25.21 | tzafrir_laptop | SQLDarkly, can you please pastebin the complete error? |
| 16:26.41 | bmoraca | my provider numbers their channels from 0 to 22 |
| 16:26.52 | bmoraca | i lose a channel if zaptel can't do the same |
| 16:27.21 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:30.36 | tzafrir_laptop | bmoraca, hmm.... those numbers are internal numbering. Why does your provider care if you call that chanel 0, 1, 314 or "first"? |
| 16:32.26 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 16:32.37 | *** join/#asterisk neurosys (n=vinix@c-71-196-8-216.hsd1.fl.comcast.net) |
| 16:33.04 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 16:34.18 | *** part/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 16:34.49 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:34.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:38.53 | bmoraca | well, because it's looking for timing slots starting at a location that zaptel aparently cannot use |
| 16:39.04 | bmoraca | as a result, I only get access to channels 1-22 now |
| 16:39.20 | bmoraca | for all intents and purposes, channel 23 doesn't exist on the provider end. |
| 16:40.19 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 16:40.40 | [TK]D-Fender | bmoraca: PASTEBIN your configs and show us the issue |
| 16:41.18 | [TK]D-Fender | bmoraca: Your provider has no clue how Zaptel/DAHDI numbers its channels for internal purposes, and this does not matter. |
| 16:41.37 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:43.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:44.13 | *** join/#asterisk maddog01 (n=minotaur@d221-91-175.commercial.cgocable.net) |
| 16:44.14 | bmoraca | all right...here's zaptel.conf and zapata.conf: http://pastebin.com/d10d13407 . The config works, except that I do not get channel 23. |
| 16:45.54 | [TK]D-Fender | bmoraca: Show us your attempt to access channel 23 |
| 16:46.07 | [TK]D-Fender | bmoraca: A failed call with PRI debug |
| 16:47.37 | rue_mohr | if I want the phone to boot at a descent speed I'll need to delete the sip.ld wont I? aka, remove after update |
| 16:48.12 | [TK]D-Fender | rue_mohr: No, phones should not touch firmware its already updated to. |
| 16:49.47 | rue_mohr | maybe its my imagination but it sure seems to take a while to boot, longer than before... hmm not sure |
| 16:49.49 | [TK]D-Fender | rue_mohr: Polycom take about 2 mins to boot |
| 16:49.53 | rue_mohr | mmm |
| 16:50.06 | [TK]D-Fender | rue_mohr: You have upgraded your firmware. Bigger to load |
| 16:50.24 | rue_mohr | [TK]D-Fender, told me to. :) |
| 16:50.41 | [TK]D-Fender | rue_mohr: Yes... and your point is? :p |
| 16:50.44 | rue_mohr | next checklist, I'm almost through 2 documents |
| 16:55.04 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 16:55.35 | *** part/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:55.47 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:56.50 | bmoraca | here's pri debug with a failed call on channel 23: http://pastebin.com/d69f78fc6 . If I change it to channel 22, it works no problem. |
| 17:00.34 | [TK]D-Fender | bmoraca: PB "zap show status" "zap show channels" |
| 17:02.28 | bmoraca | show status and show channels: http://pastebin.com/d2d4a35bb |
| 17:02.45 | SQLDarkly | OK FYI for those that have the same error. It is a semantic change in the kernel for skb_linearize(). Maybe the C can be modified by looking at old vs new. I will check and see whats up and let you all know. |
| 17:03.29 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 17:05.06 | Carlos_PHX | Anyone know if it's possible to use bindaddr= in a specific SIP account instead of general? I'm trying to differentiate calls coming from a single IP. |
| 17:06.52 | Qwell | SQLDarkly: SUSE 10? |
| 17:08.50 | russellb | Carlos_PHX: you can't ... guess you'd have to do it with iptables |
| 17:14.01 | *** join/#asterisk ttyS1 (n=julian@adsl-074-246-089-066.sip.bct.bellsouth.net) |
| 17:14.26 | *** join/#asterisk shounen_yuki (n=chatzill@63.229.69.246) |
| 17:15.16 | Katty | you know that point where you've had too much caffeine, and no matter how much caffeine you drink you still have that headache |
| 17:15.20 | Katty | that /caffeine/ headache |
| 17:15.23 | shounen_yuki | hihi ^^ new to asterisk and I just purchased a TE122 B, i would like to ask if this configuration is possible, it is not a normal config |
| 17:15.44 | ttyS1 | hello, I just installed freepbx. everything is working fine. I'm tryng to use this box to forward calls from ine carrier to another. this works but the incoming call's caleer id is not being forwarded to the outgoing carrier. instead the extension number replaces the original caller id and is then sent out to the outgoing carrier. is there anything I should configure to enable forwarding of the original caller id ? |
| 17:15.48 | Katty | which, i guess, might be a dehydration headache ^_- |
| 17:16.03 | *** join/#asterisk heison (n=heison@i209-195-69-159.cia.com) |
| 17:16.06 | jameswf | freepbx |
| 17:16.11 | jameswf | ~freepbx |
| 17:16.12 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:16.13 | ricko73 | see #freepbx |
| 17:17.03 | shounen_yuki | I want to connect a dialer diallogic t1 span card to the digium card and have the digium card emulate the telco, so that when the dailogic card dials out it is routed to a soft phone |
| 17:17.56 | shounen_yuki | for the heck of it I tired plugging in the one of the ports on the dialer to the digium card and the D channel showed active so I do have PHY |
| 17:18.30 | shounen_yuki | I did a lengthy almost 6 hour search on the forums on this and came up short |
| 17:18.59 | [TK]D-Fender | bmoraca: Do the call again and include a channel dump |
| 17:19.08 | Katty | hi fender |
| 17:19.27 | [TK]D-Fender | shounen_yuki: Sure, its can act as net or cpe |
| 17:19.34 | [TK]D-Fender | Katty: Mew. |
| 17:19.48 | bmoraca | channel dump? |
| 17:20.13 | shounen_yuki | ok so if i show it up as active with the d channel up is all i need to set is a pattern to ring to an extension |
| 17:20.14 | [TK]D-Fender | Carlos_PHX: Show us the SIP debug of the calls. |
| 17:20.25 | [TK]D-Fender | shounen_yuki: or do "whatever" |
| 17:20.40 | [TK]D-Fender | bmoraca: "core show channels concise" |
| 17:20.51 | [TK]D-Fender | bmoraca: This is interesting because it didn't even TRY the channel... |
| 17:20.55 | shounen_yuki | whatever ? |
| 17:21.49 | [TK]D-Fender | shounen_yuki: * can process the call any way you set it up to |
| 17:22.12 | [TK]D-Fender | shounen_yuki: implying "ring an extension" is restrictive |
| 17:22.21 | heison | anyone here using diamondcard.us? |
| 17:22.33 | shounen_yuki | ring in extension is exactly what i need it to do |
| 17:23.20 | shounen_yuki | is the pattern just the 10 digit +1 number of what the dilaer is trying to dial out |
| 17:25.12 | [TK]D-Fender | shounen_yuki: Well you can do whatever you want with the call once it comes in. |
| 17:25.20 | [TK]D-Fender | shounen_yuki: To * every call is jsut another call. |
| 17:25.28 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 17:25.59 | shounen_yuki | can i see a real time status on incoming calls under active channels or so forth ? |
| 17:26.05 | CrashSys | Anyone ran into an issue where when you have an agent log in with AgentCallBack it causes the queue to use all CPU when trying to ring that channel? |
| 17:26.39 | CrashSys | I actually have to go in and issue a hang-up on that channel to get it to free up |
| 17:26.50 | CrashSys | This is in 1.4.21.2 |
| 17:27.29 | shounen_yuki | also my 2nd issue is that when i plug the pri span into the telco and get a d channel and use the propper dialing rule to call out on I get no call rout, the ip phone can call other ip phones in the netowrk |
| 17:28.57 | bmoraca | [TK]D-Fender: when I make a call on channel 23, I do not get anything in show channels concise...the channel doesn't exist long enough for me to be able to do anything |
| 17:29.11 | [TK]D-Fender | bmoraca: No, do it jsut prior |
| 17:29.44 | bmoraca | still comes up blank |
| 17:30.02 | bmoraca | here's what initially turned me on to the issue: http://pastebin.com/m743d14d . channels resetting start at 1 and end at 22. |
| 17:30.19 | [TK]D-Fender | bmoraca: Ok, makes no sense... * isn't even trying to touch the channel, and you say there is nothing possibly reserving it off... |
| 17:30.37 | CrashSys | sounds like a zapata/zaptel issue |
| 17:30.38 | [TK]D-Fender | bmoraca: Checked with your telco on this? |
| 17:30.53 | CrashSys | is channel 23 defined as a B-channel? |
| 17:31.06 | shounen_yuki | ok i am going to plug our pbx into it and see if i can call into the box ^^ thx then i will do the dialer after hours, have a good one and thx |
| 17:31.07 | bmoraca | the only thing from my telco that I have is that the B-channels are to be numbered from 0 to 22 |
| 17:31.18 | *** part/#asterisk Faustov (i=fst@gentoo/user/faustov) |
| 17:31.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:32.01 | bmoraca | it's weird. channel 23 gives me congestion |
| 17:32.08 | bmoraca | and that's all |
| 17:32.24 | *** part/#asterisk shounen_yuki (n=chatzill@63.229.69.246) |
| 17:33.28 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:34.40 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 17:35.24 | CrashSys | and you can place a dial through zap channel 0? |
| 17:36.12 | [TK]D-Fender | CrashSys: *'s numbering doesn't factor in like that |
| 17:37.10 | bmoraca | yeah...if I modify zaptel.conf and zapata.conf to use 0-22, it does not fun things :P |
| 17:37.35 | Deeewayne | bmoraca: enable pri intense debug and see if you are getting a RESTART from the network |
| 17:37.37 | CrashSys | bmoraca: so you have b-chan => 1-23 in zap? |
| 17:37.43 | bmoraca | yes |
| 17:37.57 | CrashSys | is this in mexico? |
| 17:38.00 | bmoraca | no |
| 17:38.04 | CrashSys | MFCR2? |
| 17:38.08 | CrashSys | Ohh, ok... |
| 17:38.29 | *** join/#asterisk id10t_help_ (n=chatzill@76.164.167.174) |
| 17:39.39 | CrashSys | kind of weird for a PRI to start with channel 0 |
| 17:39.48 | CrashSys | shrugs |
| 17:40.43 | bmoraca | maybe i should check with digium...maybe they've got a patch for this or something |
| 17:41.12 | bmoraca | in the meantime, i'll check with the telco and see if they can modify it to use channels 1-23 instead. that should be within their ability |
| 17:42.22 | Deeewayne | bmoraca: second option is better |
| 17:42.30 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 17:44.16 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 17:44.17 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:45.01 | bmoraca | not if they're going to charge me a fee to do it |
| 17:45.13 | carrar | This is Asterisk from source? |
| 17:45.27 | carrar | (bmoraca) |
| 17:45.56 | id10t_help_ | HI, I have a ABE terminating with Digium TE122B card. I am getting garbled voice/dropped packets when I try to talk over someone while on an external call. My consultant isn't handling this fast enough so I am looking for suggestions on where to start looking for the problem? |
| 17:46.51 | Deeewayne | id10t_help_: if you have ABE you should contact Digium Support |
| 17:47.02 | Deeewayne | s/should/could |
| 17:47.30 | id10t_help_ | thought I would get a better/faster answer here. :) |
| 17:47.40 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:49.13 | Carlos_PHX | [TK]D-Fender: I think I know where your thought is going...use SIP headers in the dialplan? |
| 17:52.13 | *** join/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 17:54.37 | *** join/#asterisk jackson__ (n=jackson@68-115-83-189.dhcp.roch.mn.charter.com) |
| 17:57.50 | Deeewayne | bmoraca: AT&T TR 41459, section 3.6.5.12, note 9 states "For B-channels, the channel number equals the time slot number. The range of channel numbers supported will be 1-24" |
| 17:59.00 | carrar | HOT TALK |
| 17:59.12 | Deeewayne | I know you are using national switchtype, but I would be surprised to see the supported range differ |
| 18:00.51 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 18:04.14 | jameswf | http://www.3amsystems.com/wireline/tone-search.htm looks interesting |
| 18:05.49 | [TK]D-Fender | Carlos_PHX: If thats viable, yes. Could be the exten it targets via To;, etc |
| 18:06.16 | russellb | jameswf: I usually refer to this: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf |
| 18:11.29 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 18:12.41 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-9-29.w86-215.abo.wanadoo.fr) |
| 18:12.47 | *** part/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 18:12.55 | *** part/#asterisk id10t_help_ (n=chatzill@76.164.167.174) |
| 18:15.34 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:15.49 | flujan | ping seanbright |
| 18:16.59 | flujan | seanbright: I solved that issue with my ruby agi script. |
| 18:17.28 | flujan | the problem is that you cannot have STDIN being read inside a ruby module, you need to have it on the script asterisk calls. |
| 18:17.40 | flujan | this solved the issue I were having. thanks for your time. :) |
| 18:18.18 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-9-29.w86-215.abo.wanadoo.fr) |
| 18:18.39 | khronos | <PROTECTED> |
| 18:20.33 | *** join/#asterisk ZX81_ (i=ZX81@124.6.218.246) |
| 18:20.47 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:21.01 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 18:22.39 | [TK]D-Fender | khronos: You don't say! |
| 18:22.57 | *** join/#asterisk Get_The_Fish (n=IceChat7@75.151.94.189) |
| 18:22.59 | Deeewayne | agrees |
| 18:24.55 | *** join/#asterisk nexu (n=nexu@86.85.169.45) |
| 18:27.30 | *** join/#asterisk The_Lightside (n=Lightsid@dsl-241-98-116.telkomadsl.co.za) |
| 18:28.50 | The_Lightside | hi all, having an issue where the E1 spans are going up and down. "got a UA, but im in state 1" where do i look? |
| 18:30.26 | flujan | guys, i am trying to make asterisk dial-out using call files.. |
| 18:31.02 | flujan | I read about the failed extension an so forth... I have a dialplan like this failed,1,CDR() |
| 18:31.10 | flujan | and it is not recording on the CDR. |
| 18:31.13 | flujan | any tips? |
| 18:32.31 | The_Lightside | seems like majority of the irc population are away.... :) |
| 18:33.48 | manxpower | flujan: where did you read about that failed extension? |
| 18:36.10 | flujan | manxpower: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out#Thefailedextension |
| 18:36.44 | manxpower | I sort of thought it would be part of the official docs |
| 18:39.20 | *** join/#asterisk ZX81 (i=ZX81@124.6.218.246) |
| 18:39.31 | flujan | manxpower: when i dial-out using call files asterisk do not generate debug info in the cli. I can turn this on? Or even get the CDR result of unaswered calls? I did set the cdr.conf unaswered=yes but no luck |
| 18:40.02 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:41.27 | manxpower | flujan: use chan_local so everything goes thru the dialplan |
| 18:42.13 | flujan | i am using the SIP/11111/44489652 i must use Local/SIP/111111/44489652 to make it work? |
| 18:42.28 | manxpower | flujan: looks like you need to read localchannel.txty |
| 18:42.32 | manxpower | and localchannel.txt |
| 18:42.49 | *** join/#asterisk ZX81_ (n=matt@202.20.97.211) |
| 18:42.55 | flujan | manxpower: sorry, never read them will do it now ... thanks for the help manxpower |
| 18:47.03 | *** join/#asterisk pecanha (n=e@189.106.100.128) |
| 18:48.44 | codefreeze-lap | flujan: Someone else had problems that way, call files not getting recorded right... look thru the bug 14167 and see if it matches your situation |
| 18:49.12 | flujan | codefreeze-lap: ok thanks for the tip. |
| 18:50.03 | pecanha | hello guys, can anyone recommend few sites with dialplans available? |
| 18:50.54 | Carlos_PHX | [TK]D-Fender: If you have a sec, would you look at this code and see what you think? http://televolve.pastebin.com/m5504fe7c |
| 18:53.32 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:53.51 | flujan | codefreeze-lap: yep. it matchs my bug here... i will give a try generating the calls using AMI originate. |
| 18:54.28 | *** join/#asterisk `paul (n=kutimoy@121.127.6.131) |
| 18:55.02 | [TK]D-Fender | Carlos_PHX: Now show me a call going through it |
| 18:55.07 | `paul | if i set a variable in originate api funtion how can i access it inside a context? |
| 18:55.40 | codefreeze-lap | flujan: I investigated a similar problem with call files on a different tree, and have a patch, but it may not be the same prob. |
| 18:57.03 | Carlos_PHX | [TK]D-Fender: Can't do it yet, production system. |
| 18:57.05 | The_Lightside | does anybody have any idea as to my problem posted a few mins ago? |
| 18:57.13 | Carlos_PHX | Will have to wait until after 9pm. |
| 18:57.38 | Carlos_PHX | Or until the provider points a number to my test server, whichever comes first. |
| 18:58.09 | *** join/#asterisk xkr47 (i=xkr47@a88-112-18-173.elisa-laajakaista.fi) |
| 18:58.14 | xkr47 | yello! |
| 19:02.54 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 19:03.00 | *** part/#asterisk mtutaj (n=mtutaj@76-231-68-228.lightspeed.cicril.sbcglobal.net) |
| 19:06.29 | *** join/#asterisk [netman] (n=netman@70.Red-88-25-137.staticIP.rima-tde.net) |
| 19:15.38 | bmoraca | ok, now that's just weird. channel 23 just started working again. |
| 19:17.10 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 19:20.26 | *** part/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 19:26.03 | *** join/#asterisk saftsack (n=oliver@g227085175.adsl.alicedsl.de) |
| 19:28.13 | *** join/#asterisk dieguito84 (n=diego@host204-186-dynamic.23-79-r.retail.telecomitalia.it) |
| 19:29.43 | flujan | codefreeze-lap: i will keep track of this bug...I will report further information. |
| 19:30.01 | *** join/#asterisk thetrooper7 (n=thetroop@adsl56-150.kln.forthnet.gr) |
| 19:30.20 | codefreeze-lap | flujan: good... let's see if we can close it soon. |
| 19:30.37 | thetrooper7 | Hello ppl where can i ask about a problem with ASterisk? Am i in the right room? |
| 19:30.47 | Qwell | ~help |
| 19:30.53 | Qwell | ~ask |
| 19:30.54 | jbot | it has been said that ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:30.57 | Qwell | glares at jbot |
| 19:31.06 | thetrooper7 | ~help |
| 19:31.19 | russellb | this is the right room. |
| 19:31.50 | thetrooper7 | :) |
| 19:31.58 | *** join/#asterisk dec (n=tom@unaffiliated/dec) |
| 19:32.32 | *** join/#asterisk reneger (n=reneger@dslb-088-078-119-125.pools.arcor-ip.net) |
| 19:32.40 | thetrooper7 | ~help I setup Up AsteriskWin32, all users register OK but when then is initiated,there is no Voice transfered... |
| 19:32.49 | thetrooper7 | i have open the ports... |
| 19:34.02 | Qwell | asteriskwin32 does not exist. |
| 19:34.10 | Qwell | Also, this is not #asteriskwin32 |
| 19:35.22 | thetrooper7 | i got ot go.. i'll come back tomorrow to ask what i want.. |
| 19:35.25 | thetrooper7 | thx anyway |
| 19:35.29 | russellb | heh |
| 19:35.33 | flujan | lol |
| 19:35.47 | flujan | this is sick to use asterisk on windows lol |
| 19:35.47 | thetrooper7 | i have a bus to catch :) |
| 19:36.03 | thetrooper7 | yes but i have no other choice.. |
| 19:36.07 | Qwell | thetrooper7: The software you installed does not exist. You can't get help with it here. |
| 19:36.11 | thetrooper7 | watever.. |
| 19:36.14 | thetrooper7 | night night |
| 19:36.38 | *** join/#asterisk Ast001 (n=urke@cable-89-216-157-103.dynamic.sbb.rs) |
| 19:36.43 | Qwell | right |
| 19:36.59 | Qwell | sets +r on other asterisk channels as well |
| 19:38.54 | Ast001 | hi I have problem with Asterisk 1.4.21. When my operators listen music on hold sound is cut. (with breaks) .Its like moh is trembeling. |
| 19:39.21 | Ast001 | and there is no lost packets,download/upload is ok |
| 19:40.13 | Ast001 | there is 6 points for packages when they travel form server to operator for 102ms |
| 19:40.37 | ZX81 | run mtr between sites |
| 19:40.47 | ZX81 | ms is not as important as jitter |
| 19:41.03 | ZX81 | and zttest -vvv |
| 19:41.08 | ZX81 | aiming for 100% |
| 19:41.28 | ZX81 | and is moh coming from files or stream |
| 19:41.38 | Ast001 | moh is coming from files |
| 19:42.00 | ZX81 | do you get the same on calls or only on moh |
| 19:42.06 | Ast001 | only on moh |
| 19:42.22 | ZX81 | where are the moh files from? |
| 19:42.34 | Ast001 | from server's /var/lib/asterisk/moh |
| 19:42.36 | ZX81 | supplied with * |
| 19:42.37 | ZX81 | ? |
| 19:43.07 | ZX81 | what do you get with zttest -vv? |
| 19:43.13 | Ast001 | wait a sec |
| 19:43.16 | ZX81 | kk |
| 19:43.31 | Ast001 | 100% |
| 19:43.40 | Ast001 | 99.9998 |
| 19:43.46 | ZX81 | yeah that's fine |
| 19:43.56 | ZX81 | silence suppression disabled on phones? |
| 19:44.02 | Ast001 | i use xlite |
| 19:44.12 | ZX81 | xlite == phone then :) |
| 19:44.35 | Ast001 | gsm codec |
| 19:44.39 | ZX81 | think it's called conserve bandwidth |
| 19:44.41 | ZX81 | sec |
| 19:44.44 | ZX81 | loading eyebeam |
| 19:44.59 | Ast001 | Use auto gain control ? |
| 19:45.12 | Ast001 | noise reduction ? acustic echo cancellation ? |
| 19:45.21 | ZX81 | options -> advanced -> Network -> Preserve bandwidth during silence periods |
| 19:45.29 | ZX81 | should be disabled |
| 19:45.45 | Ast001 | i Disabled it |
| 19:45.53 | Ast001 | Preserve bandwith duriing silence periods |
| 19:45.55 | ZX81 | was disabled before or you disabled it now? |
| 19:45.56 | Ast001 | is disabled |
| 19:46.05 | Ast001 | it was disabled |
| 19:46.15 | Ast001 | I tried to enable it and got same problem |
| 19:46.22 | ZX81 | and you can reproduce the problem with the soft phone you are using? |
| 19:46.31 | *** join/#asterisk RoyK (n=roy@ip-244-29-149-91.dialup.ice.no) |
| 19:46.37 | Ast001 | problem is still here |
| 19:46.46 | Ast001 | I think it might be conected with jiter bufer |
| 19:46.46 | ZX81 | inside /etc/asterisk/asterisk.conf there is an option internaltiming |
| 19:47.11 | ZX81 | under [options] |
| 19:47.15 | ZX81 | what is this set to? |
| 19:47.34 | Ast001 | to yes |
| 19:47.38 | ZX81 | ok |
| 19:47.38 | Ast001 | but it is commented |
| 19:47.42 | ZX81 | :D |
| 19:48.05 | ZX81 | ok so, a couple of things, try disable jitterbuffer |
| 19:48.05 | RoyK | hi. I've heard some telco companies in the US are charging for non-answered calls, as in starting charging from the ring indicator. Is it true they are doing this? legally? |
| 19:48.05 | Ast001 | should I uncoment it ? |
| 19:48.15 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.202.100) |
| 19:48.23 | ZX81 | Ast001 yeah |
| 19:48.33 | ZX81 | you'll probably need to restart asterisk |
| 19:48.40 | ZX81 | make sure [options] is uncommented too |
| 19:48.40 | Ast001 | and to disable jiterbuffer in sip.conf ? |
| 19:48.46 | ZX81 | one thing at a time |
| 19:48.56 | ZX81 | but if this doesn't work, change it back and then try jb |
| 19:49.09 | Ast001 | ok I'll do it |
| 19:49.44 | RoyK | Ast001: I know a little about the jb, but didn't get the problem - may you repeat it, please? |
| 19:49.54 | ZX81 | call breakup in moh |
| 19:50.06 | ZX81 | silence suppression disabled in soft phone |
| 19:50.07 | RoyK | how can that be related to the jb? |
| 19:50.16 | Ast001 | My operators working on xlite sip phone are experiancing bad moh quality |
| 19:50.16 | ZX81 | 100ms ping between sites |
| 19:50.22 | Ast001 | cut voice trablming voice |
| 19:50.24 | RoyK | the jb is rtp only, shouldn't matter |
| 19:50.31 | Ast001 | only on moh |
| 19:50.37 | Ast001 | when they talk it is ok |
| 19:50.41 | RoyK | without a jb, the audio will suck bug time |
| 19:50.44 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 19:50.49 | ZX81 | :D bug time |
| 19:50.52 | Ast001 | it looks better now |
| 19:50.59 | ZX81 | type bug more than big :D |
| 19:51.00 | Ast001 | when I enabled options |
| 19:51.02 | RoyK | Ast001: try disabling rtptimeout if it's set |
| 19:51.03 | pecanha | guys, how can I put an mp3 file to be played by asterisk? I mean, I know how can I play a file, I just want to use mp3. |
| 19:51.29 | RoyK | Ast001: also, after how long is the call disconnected? |
| 19:51.30 | Qwell | pecanha: asterisk-addons has format_mp3, but I would suggest converting them to wav. |
| 19:51.31 | Ast001 | it looks much better now |
| 19:51.52 | ZX81 | cool |
| 19:51.59 | Ast001 | it sounds great now |
| 19:52.09 | mvanbaak | if you want a compressed one use ogg |
| 19:52.12 | Ast001 | thanksZX81 |
| 19:52.16 | ZX81 | no probs man |
| 19:52.16 | RoyK | the original jb code submitted to 1.2 didn't have that problem :P |
| 19:52.26 | Ast001 | is ZX81 from ZX81 processor ? |
| 19:52.36 | ZX81 | :D yah |
| 19:52.43 | RoyK | Ast001: he's still running that with asterisk :D |
| 19:52.48 | ZX81 | heh yeah |
| 19:52.51 | pecanha | Qwell: wav works by default? |
| 19:52.53 | RoyK | ZX81: long time |
| 19:52.55 | ZX81 | .00025 channels |
| 19:52.58 | Ast001 | I used ZX Spectrum long time ago :) Ser Clive Sinclair rules |
| 19:53.16 | pecanha | I mean, just put the wav on folder sounds? |
| 19:53.17 | ZX81 | RoyK: I know - I tend to only come online when I'm fixing bugs |
| 19:53.19 | ZX81 | :D |
| 19:53.44 | RoyK | ZX81: I tend to go into #asterisk if I need to ask some telecom questions - not so often these days |
| 19:53.54 | ZX81 | yah |
| 19:53.55 | mvanbaak | pecanha: yeah, wav works by default |
| 19:55.01 | RoyK | Ast001: what did you do? swich to wav or disable the jb? |
| 19:55.13 | Ast001 | I just did what ZX81 told me |
| 19:55.18 | Katty | had a nap |
| 19:55.21 | Ast001 | enable options if asterisk.conf |
| 19:55.27 | Ast001 | in asterisk.conf |
| 19:55.28 | RoyK | Ast001: which ones? |
| 19:55.32 | Ast001 | and internal=yes |
| 19:56.01 | Ast001 | internal_timing=yes |
| 19:56.21 | RoyK | does asterisk support posix timers these days? |
| 19:56.34 | mvanbaak | yeah |
| 19:56.48 | mvanbaak | erm |
| 19:56.52 | mvanbaak | hang on |
| 19:56.55 | Ast001 | well in my region villagers has wiseman say "Feed pigs and don't touch anything" |
| 19:56.58 | mvanbaak | we do pthread and timerfd |
| 19:57.03 | Ast001 | when it works |
| 19:57.05 | Ast001 | :) |
| 19:57.34 | pecanha | I need to use: exten => s,n,Playback(filp) or exten => s,n,Playback(filp.wav) ? |
| 19:57.57 | Ast001 | now everything is fine thanks to ZX81 :) It is pleasure to hear moh now :) |
| 19:58.06 | Ast001 | see you after friends |
| 19:58.32 | RoyK | pecanha: the extension is added automatically |
| 19:58.36 | *** part/#asterisk Ast001 (n=urke@cable-89-216-157-103.dynamic.sbb.rs) |
| 19:59.38 | pecanha | I'll try, to reload extensions.conf, will asterisk -rx reload work? |
| 20:01.34 | *** join/#asterisk Daejeo (n=chatzill@116.126.121.31) |
| 20:02.15 | ZX81 | pecanha: dialplan reload |
| 20:02.16 | Daejeo | is it possible to use a722 with *1.4? |
| 20:02.18 | ZX81 | g722? |
| 20:02.18 | ZX81 | :) |
| 20:02.18 | Daejeo | is it possible to use g722 with *1.4? |
| 20:02.28 | Daejeo | typo |
| 20:02.37 | ZX81 | no translation |
| 20:02.42 | pecanha | unfortunatelly, it didn't work :( can't hear any sound with my .wav |
| 20:02.44 | pecanha | any idea? |
| 20:02.47 | ZX81 | asterisk -rx 'show translation' |
| 20:03.07 | Daejeo | * 1.6 ok? |
| 20:03.12 | ZX81 | pecanha: file filp.wav |
| 20:03.39 | pecanha | filp.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz |
| 20:03.43 | Daejeo | freeswitch |
| 20:03.46 | Daejeo | :) |
| 20:04.09 | ZX81 | Daejeo: 1.6 = yes |
| 20:04.27 | ZX81 | you can check yourself with "core show translation" |
| 20:04.32 | pecanha | ZX81: is that right, correct? |
| 20:05.13 | ZX81 | pecanha: I'd do 16 bit, mono, 8000hz |
| 20:05.52 | pecanha | ZX81: that's why I can't hear? or is something else? |
| 20:06.00 | ZX81 | dunno :) |
| 20:06.06 | ZX81 | can you hear other files? |
| 20:06.09 | pecanha | yes |
| 20:06.21 | pecanha | didn't try other wav files however |
| 20:06.24 | ZX81 | I personally use Steinberg's Wavelab to convert all files to alaw |
| 20:06.43 | ZX81 | or you could use sox to convert to sln |
| 20:07.16 | ZX81 | I use |
| 20:07.23 | Daejeo | ZX81: does it ship with *1.6? |
| 20:07.32 | ZX81 | sox infile.wav -t raw -r 8000 -s -w -c 1 outfile.sln |
| 20:07.37 | ZX81 | Daejeo: yeah |
| 20:08.26 | ZX81 | I'm not personally running 1.6 except in the lab - so mine should be a pretty stock 1.6 |
| 20:08.45 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:09.00 | pecanha | hmm I don't have sox |
| 20:09.09 | ZX81 | apt-get install sox |
| 20:09.11 | ZX81 | or |
| 20:09.14 | ZX81 | yum install sox |
| 20:09.32 | ZX81 | :) or yum install debian |
| 20:09.33 | ZX81 | :D |
| 20:09.39 | pecanha | ok :p |
| 20:09.43 | ZX81 | :D |
| 20:11.00 | pecanha | wow yum couldn't find libvorbis mirror :/ |
| 20:11.04 | pecanha | its a dependecy |
| 20:11.22 | pecanha | I'll try to install from source |
| 20:11.25 | ZX81 | yum update |
| 20:11.29 | ZX81 | maybe |
| 20:12.47 | Qwell | pecanha: don't do that |
| 20:13.01 | *** part/#asterisk RoyK (n=roy@ip-244-29-149-91.dialup.ice.no) |
| 20:13.05 | Daejeo | ZX81: no audio with *1.4 |
| 20:14.03 | ZX81 | maybe you have a dial command which requires transcoding? |
| 20:14.24 | pecanha | Qwell: why? |
| 20:14.30 | ZX81 | just do exten => 1234,1,Dial(SIP/otherone) |
| 20:14.53 | Qwell | pecanha: Installing dependencies from source like that on a package-based distro is never a good idea. |
| 20:15.18 | Qwell | If it's just a mirror issue, that can be easily worked around. Either try again (possibly several times) or download the package from the vendor. |
| 20:16.12 | ZX81 | in debian doing an apt-get update updates the urls as well |
| 20:16.22 | Daejeo | ZX81: anyway to use g722 on *1.4? |
| 20:16.26 | ZX81 | I don't remember if yum update does the same |
| 20:16.37 | ZX81 | Daejeo: yeah only between devices though |
| 20:16.41 | ZX81 | no message playback |
| 20:16.46 | ZX81 | unless they are in g722 |
| 20:16.50 | Qwell | ZX81: most distros use yum-fastestmirror, which will (usually) change URLs each time you try to install something |
| 20:16.57 | ZX81 | right |
| 20:16.57 | Daejeo | ah ok |
| 20:17.03 | Daejeo | only pass thru? |
| 20:17.06 | ZX81 | yah |
| 20:17.09 | Qwell | if not, then it's probably pointing to a round-robin resolver or something |
| 20:17.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 20:17.36 | ZX81 | :) or add a nonexistant hosts entry for the server that bombed :D |
| 20:17.58 | ZX81 | tzafrir_laptop: hey that machine at the hotel restarted no problem - thanks for that fix |
| 20:18.06 | Daejeo | do you have any phone with g722 codec |
| 20:18.19 | *** join/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
| 20:18.20 | Daejeo | polycom? |
| 20:18.22 | ZX81 | probably the polycom in reception |
| 20:18.28 | ZX81 | but I'm at home at the mo |
| 20:18.43 | Daejeo | i just got the codec |
| 20:18.58 | Daejeo | i have polycom 501 |
| 20:20.16 | ipso | I just discovered that my Asterisk server is apparently wide open to allow remote SIP connections to dial through it, and of course its being abused. What settings are required to insist that Asterisk only accepts calls/registrations with usernames/passwords? |
| 20:20.16 | ZX81 | ipso: make a guest account with context=fckoff |
| 20:20.16 | ZX81 | or |
| 20:20.16 | ZX81 | allowguest=no |
| 20:20.16 | ZX81 | there's something like that in the sample config |
| 20:20.17 | Qwell | or both :D |
| 20:20.17 | ZX81 | :D |
| 20:20.45 | Daejeo | how to : polycom -codec update? |
| 20:20.58 | ipso | ZX81: wtf, thats enabled by default? |
| 20:20.58 | ZX81 | :) join #polycom ? |
| 20:21.08 | ZX81 | ipso yeah think so |
| 20:21.12 | ZX81 | with context of default |
| 20:21.19 | ipso | ZX81: uhg, thats nuts. |
| 20:21.24 | ZX81 | normally [default] just contains demo etc |
| 20:22.10 | ZX81 | read /usr/src/asterisk/doc/security.txt |
| 20:22.50 | ZX81 | actually nothing in 1.4 about that :) |
| 20:23.11 | ZX81 | configs/sip.conf.sample has: |
| 20:23.14 | ZX81 | ;allowguest=no ; Allow or reject guest calls (default is yes) |
| 20:23.19 | bmoraca | ipso, in your sip.conf in the same place you would specify your local network, etc, you can specify a context for unregisteres sip connections |
| 20:24.08 | ipso | Anyone know Chinese, I have a few hours of recorded calls from these people making expensive calls through my Asterisk server. :( |
| 20:24.30 | bmoraca | you're probably not going to get that back |
| 20:25.24 | ZX81 | we had a period where people were paying for credit with stolen paypal accounts |
| 20:25.31 | *** join/#asterisk jshriver (n=jshriver@72.240.39.37) |
| 20:25.34 | jshriver | greetings |
| 20:25.39 | ZX81 | heya |
| 20:25.46 | jshriver | what is the diff in geni586, net4801, etc, etc |
| 20:25.55 | ZX81 | for astlinux |
| 20:25.56 | jshriver | there's no readme, guessing geni586 is a generic |
| 20:26.00 | jshriver | yup |
| 20:26.02 | ipso | bmoraca: Yeah, its obviously my fault... Would just be curious to know what they are talking about. |
| 20:26.06 | ZX81 | i586 works on normal pc |
| 20:26.13 | ZX81 | 4801 is for soekris 4801 |
| 20:26.19 | ZX81 | 5501 is for soekris 5501 |
| 20:26.42 | ZX81 | the 4801 has been end of lifed |
| 20:26.47 | jshriver | ok have a question, if I try the newest version 0.6, I rtc: lost some interrupts at 1024Hz, or a Kernel panic when trying to dd /cdrom/root.squafs.. |
| 20:26.47 | ZX81 | replaced by the 5501 |
| 20:26.48 | bmoraca | ipso, under the general context in sip.conf, specify the following: "context=sip-unregistered". then in extensions.conf, create a context sip-unregistered and do a _.,1,Hangup() or something like that |
| 20:27.03 | jshriver | if I try version 0.4.8 it has an old driver for r8168 and doesnt work |
| 20:27.14 | ZX81 | jshriver: probably best join #astlinux and trying to find darrick |
| 20:27.26 | jshriver | hrm will try again noone there really been checking days now |
| 20:27.29 | jshriver | appreciate it though |
| 20:27.30 | pecanha | ZX81: do you recommend a good tutorial/book over dialplan? :) |
| 20:27.42 | ZX81 | jshriver: there was a fix for rtc a couple of days ago |
| 20:27.52 | ZX81 | pecanha: yeah lmadsen's book |
| 20:27.54 | ZX81 | ~tfot |
| 20:27.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 20:28.04 | pecanha | ZX81: already read it hehe |
| 20:28.10 | *** join/#asterisk zamba (i=marius@sveigde.hih.no) |
| 20:28.15 | ZX81 | jshriver: there's also a pretty active mailing list |
| 20:28.21 | zamba | is it possible to establish gsm calls through asterisk? |
| 20:29.00 | zamba | if you have a sim card and some radio/antenna device or a mobile phone attached? |
| 20:29.11 | ZX81 | brb meeting |
| 20:32.30 | bmoraca | ipso, what kind of trunking do you have if you don't mind me asking? |
| 20:34.35 | *** join/#asterisk susinths (n=susinths@s0021-0018.dsl.start.no) |
| 20:34.42 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:35.30 | jshriver | ty :) have a good day everyone |
| 20:35.35 | *** join/#asterisk citywok (n=chatzill@67-148-102-20.dia.static.qwest.net) |
| 20:37.07 | *** join/#asterisk wildzero-cw (n=chatzill@213.216.10.219) |
| 20:38.06 | citywok | I have a dialplan with quite a few lines for my outbuond dialing of 11digit numbers, i want to send only 800#'s to a different provider, but apply all the generic rules from the dialplan to the call, what is the easiest way to do this, without having to copy/paste a bunch of lines? |
| 20:38.06 | ipso | bmoraca: Nothing fancy at all, its just for personal/home use |
| 20:38.07 | *** part/#asterisk wildzero-cw (n=chatzill@213.216.10.219) |
| 20:41.00 | bmoraca | well, in that case, you might consider not exposing the sip/rtp ports to the internet...then you don't have to worry |
| 20:41.36 | *** join/#asterisk eliyahud (n=eliyahud@84.94.66.209.cable.012.net.il) |
| 20:46.42 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 20:50.42 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:e405:8332:ecdc:a26a) |
| 20:51.03 | s34n | anybody have a good experience with a voip intercom? |
| 20:52.02 | *** join/#asterisk marv0997 (n=marv@205.211.247.62) |
| 20:52.09 | *** part/#asterisk marv0997 (n=marv@205.211.247.62) |
| 20:52.24 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 20:53.37 | *** join/#asterisk Charlie77 (n=tekach@89-212-30-165.dynamic.dsl.t-2.net) |
| 20:53.41 | Charlie77 | hi there :-) |
| 20:53.45 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 20:53.54 | Charlie77 | anyone alive? |
| 20:54.49 | Corydon76-dig | ~ask |
| 20:54.50 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:57.04 | Charlie77 | :-) |
| 20:57.05 | carrar | I'm Alive!!! |
| 20:58.47 | drmessano | OMG CARRAR |
| 20:58.51 | drmessano | You're ok!!! |
| 20:58.57 | drmessano | sobs |
| 20:58.59 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 20:59.08 | carrar | O |
| 20:59.08 | drmessano | I told them not to give up! |
| 20:59.08 | carrar | M |
| 20:59.09 | carrar | G |
| 20:59.45 | Charlie77 | I'm using Asterisk 1.4.22, Freepbx 2.5.1.1 with Eicon Diva PRI and chan_capi. I'd like to achieve the following functionality: |
| 20:59.46 | drmessano | I know I never told you this before.. but the thought that I had lost you made me realize just how much I love you!!! |
| 20:59.52 | drmessano | YES, I LOVE YOU!!!!! |
| 21:00.03 | drmessano | breaks down in moviemonious glory |
| 21:00.04 | Charlie77 | Customer calls - Message is played back: "Please dial your account code" - Customer dials his account code - Another message is played back: "Thank you. Now dial your meter readings (status)" - Customer dials his meter reading (status of the gas /water meter) |
| 21:00.07 | carrar | Try #freepbx |
| 21:00.31 | carrar | hahha |
| 21:00.37 | drmessano | Wait.. You're "carrar" |
| 21:00.40 | drmessano | Not carar |
| 21:00.42 | carrar | I had to read that twice |
| 21:00.45 | drmessano | Oh shit.. I hate you |
| 21:00.52 | drmessano | I miss carar :( |
| 21:00.57 | Charlie77 | Dialed DTMF numbers should be fetched and saved to MySQL table in format "cid|account code|meter reading". |
| 21:01.09 | Charlie77 | this is a question for #freepbx? |
| 21:01.27 | carrar | You can create that dialplan yourself |
| 21:01.38 | carrar | make sure you are running asterisk from source |
| 21:01.44 | Charlie77 | I am. |
| 21:02.05 | carrar | You mentioned freepbx |
| 21:02.09 | carrar | whats that about |
| 21:02.23 | Charlie77 | I am offering free beer and ethernal friendship for a hint on how to achieve that |
| 21:02.24 | Charlie77 | :) |
| 21:02.38 | drmessano | Im not into ethernal friendships |
| 21:02.41 | drmessano | I dont swing that way |
| 21:02.45 | beek | Charlie77: Look up func_odbc |
| 21:02.46 | carrar | You need to learn how to create dialplans |
| 21:02.56 | tzafrir_laptop | ethernet friendship? |
| 21:02.58 | carrar | I would write a AGI |
| 21:03.01 | bmoraca | using AGI would probably be the simplest...or the built-in ODBC functions. |
| 21:03.07 | drmessano | tzafrir_laptop for the WiN |
| 21:03.09 | carrar | gmta |
| 21:03.22 | Charlie77 | gents, I have to tell you that I am not an expert |
| 21:03.22 | drmessano | "Ethernet friendship" |
| 21:03.25 | drmessano | thats funny |
| 21:03.26 | bmoraca | you cannot use freepbx to create the application, though. that will have to be done in custom |
| 21:03.34 | Charlie77 | but putting something together will save me 500$ |
| 21:03.34 | drmessano | "I will be YOUR CDP neighbor" |
| 21:03.35 | carrar | Charlie77, no one is when they start |
| 21:03.44 | carrar | Think of all the fun stuff you get to learn |
| 21:03.45 | drmessano | "I will be your failover node" |
| 21:04.08 | ricko73 | drmessano: is that akin to "I'll be you're Huckleberry" |
| 21:04.15 | drmessano | I almost made a master slave replication joke, but this is a family channel |
| 21:04.52 | tzafrir_laptop | Charlie77, you haven't actually asked your question |
| 21:04.55 | Charlie77 | drmessano: ethernal friendship was a joke of course. beer was not :-) |
| 21:05.14 | edoceo | If I buy a DID from didx how do I plug it into my * system? Do I still need to work with a local carrier? |
| 21:05.24 | drmessano | Charlie77: I am still trying to figure out what an "ethernal friendship" is.. It sounds.. kinky |
| 21:05.43 | drmessano | or involves lots of CAT5e |
| 21:05.46 | ricko73 | in the immortal words of H. Simpson "don't toy with me woman" (when Marge said "we're all out of beer") |
| 21:05.46 | bmoraca | edoceo, you'll still need a local carrier for origination, yes |
| 21:05.56 | Charlie77 | tzafrir_laptop: is that possible to achieve? any hint or solution would be highly appreciated |
| 21:06.07 | bmoraca | termination will come through a SIP trunk to DIDX |
| 21:06.27 | edoceo | bmoraca: so that means the side that's calling me right? |
| 21:06.27 | carrar | Charlie77, yes you can do that in asterisk |
| 21:06.27 | eppigy | HELLO |
| 21:06.30 | drmessano | ricko73: I loved when they were gonna order a pizza.. and it was programmed on the "fire" speeddial button on their phone |
| 21:06.31 | eppigy | cap |
| 21:06.33 | eppigy | s |
| 21:06.34 | eppigy | hi |
| 21:06.49 | ricko73 | lol |
| 21:06.53 | bmoraca | yes. when someone calls you, that gets to you through a SIP trunk established with DIDX. |
| 21:06.58 | tzafrir_laptop | Charlie77, surely can be done with a simple AGI script |
| 21:07.08 | tzafrir_laptop | Not sure if it can be sanely done with pure dialplan |
| 21:07.10 | drmessano | Makes me want to program 911 to call Papa Johns on my PBX at home |
| 21:07.28 | eppigy | until on day you are at work |
| 21:07.33 | eppigy | and dial without thinking |
| 21:07.39 | eppigy | *one |
| 21:07.39 | drmessano | Hell, at leasy Papa Johns comes on time |
| 21:07.43 | eppigy | lol |
| 21:07.44 | eppigy | true |
| 21:07.55 | drmessano | I call for a deputy here and its like 2 hours |
| 21:07.59 | bmoraca | however, when you call someone else, that will have to be done through your own local carrier or another SIP provider that allows for origination |
| 21:08.15 | drmessano | Better off calling the local news media/ambulance chasers |
| 21:08.25 | drmessano | Maybe they can help and get an AP award |
| 21:08.59 | drmessano | Ok, need to go to the pharmacy and see if they got MAH DOPE |
| 21:09.11 | drmessano | I dont wanna haf to slap a ho |
| 21:09.14 | edoceo | bmoraca: thanks! |
| 21:09.43 | bmoraca | NP |
| 21:09.52 | bmoraca | DIDX is a very intriguing service |
| 21:10.12 | Charlie77 | tzafrir_laptop: is that AGI script simple enough to be posted here -- I will try to make it work, I just need something to start with |
| 21:11.30 | bmoraca | he'd have to write it first. some people make their livings writing custom asterisk scripts. |
| 21:11.58 | *** join/#asterisk Micho123 (i=mcho123@80.77.180.4) |
| 21:12.42 | Micho123 | hi all,trying to install add-ons on asterisk but getting the following error during make...http://pastebin.com/d496e09d2 |
| 21:13.19 | Qwell | Micho123: Looks like you aren't installing the correct version. What version of asterisk and what version of asterisk-addons? |
| 21:13.28 | Charlie77 | bmoraca: I am aware of that. my question was humble and respectful. If that's complicated piece of code, I'll rent a coder. I am willing to pay a reasonable fee. |
| 21:13.53 | Micho123 | Charlie77, 1.4.22.1 |
| 21:14.01 | Qwell | Micho123: well, there you go. |
| 21:14.06 | Qwell | 1.4 != 1.2 |
| 21:14.20 | Micho123 | Qwell, thanks a lot |
| 21:14.26 | pecanha | ZX81: it worked after sox :) |
| 21:15.45 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:17.08 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 21:21.12 | bmoraca | Charlie, depending on how much time you have until the project needs to be completed, it might be very beneficial to attempt the project yourself |
| 21:21.22 | Charlie77 | I agree |
| 21:21.31 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:287a:d37a:e462:b170) |
| 21:21.34 | bmoraca | if you need it done very quickly, someone who has experience doing that type of thing would be the better option |
| 21:21.45 | bmoraca | who set your asterisk system up? |
| 21:22.10 | Charlie77 | I did all that. I also did some simple custom dialplans. |
| 21:22.41 | bmoraca | well, then you've probably got a fair foundation |
| 21:22.48 | bmoraca | do you have any other programming experience? |
| 21:24.24 | Charlie77 | not at all, unfortunately. |
| 21:24.56 | bmoraca | that might make it a little difficult then |
| 21:25.08 | bmoraca | you're probably going to want to use PHP as your AGI language |
| 21:25.21 | bmoraca | although, you could look at using Asterisk's internal ODBC functions |
| 21:26.09 | *** join/#asterisk linuxvoip (n=susinths@s0021-0018.dsl.start.no) |
| 21:26.54 | Micho123 | Hi all, After I install addons to asterisk and create asterisk database for savind CDRs inside where I can configure database connection?I need to save data in a mysql server located on another server...Other than asterisk |
| 21:27.03 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 21:29.10 | Charlie77 | Micho123: cdr_mysql.conf in your /etc/asterisk directory |
| 21:29.57 | *** join/#asterisk nix8n82 (n=nate@63.162.26.149) |
| 21:30.17 | *** join/#asterisk marv0997 (n=marv@205.211.247.62) |
| 21:30.28 | CrashSys | Anyone ever has Asterisk lock-up after an agent logs into a queue using agent call-back login? |
| 21:30.29 | *** part/#asterisk dec (n=tom@unaffiliated/dec) |
| 21:30.30 | Charlie77 | bmoraca: do you have a knowledge and will to do this against payment? |
| 21:31.03 | Micho123 | Charlie77, cannot find this file inside my asterisk folder |
| 21:33.06 | Charlie77 | did you install asterisk addons? |
| 21:33.13 | Micho123 | Charlie77, yes |
| 21:33.21 | Micho123 | Charlie77, version 1.2 |
| 21:33.23 | Charlie77 | with mysql option? |
| 21:34.00 | Micho123 | Charlie77, ah no...Just tar-zxvf then make and make install |
| 21:34.46 | Charlie77 | try |
| 21:34.48 | Charlie77 | make menuconfig |
| 21:35.01 | Charlie77 | and then select mysql option under second menu (cdr) |
| 21:35.03 | Charlie77 | :) |
| 21:35.04 | Micho123 | Charlie77, ok |
| 21:35.51 | Charlie77 | but i am not sure if this works for 1.2 --> it works for 1.4 |
| 21:36.36 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 21:38.04 | bmoraca | Charlie77 unfortunately, I don't really have the time to do that. |
| 21:38.29 | bmoraca | if you search on voip-info.org, you'll find examples of 90% of what you need to create your solution |
| 21:39.13 | Charlie77 | bmoraca: thank you anyway for pointing me into correct direction. |
| 21:42.29 | korihor | hi :) how pause realtime agents? |
| 21:43.16 | korihor | PauseQueueMember don't work for realtime agents :( |
| 21:43.30 | Linuturk | I have a pri coming in with 24 channels, and had DID's for fax lines about the office. In zapata.conf, the fax lines are configured as channels 73-84. There is a analog channel bank connected to the asterisk server where all the fax machines plug in. Where would I find out what channels on the PRI actually carry the fax DID's so I can turn off echo cancellation in our echo canceller? |
| 21:44.59 | Linuturk | for those specific channels* |
| 21:45.10 | bmoraca | Linuturk, depending on how your dialplan is set up, it's probably random |
| 21:46.57 | Micho123 | Charlie77, XXX near mysqladd on on menuselect...this means there is a conflict...ehat should I make here? |
| 21:48.15 | drmessano | Look down below |
| 21:48.18 | drmessano | It tells you |
| 21:48.21 | drmessano | Requires: |
| 21:49.32 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 21:50.19 | Charlie77 | Micho123: do you have mysql mysql-server mysql-devel installeD? |
| 21:50.46 | Micho123 | yes |
| 21:50.51 | Micho123 | Charlie77, yes |
| 21:53.19 | Charlie77 | okay, lets try this |
| 21:53.22 | Charlie77 | make clean |
| 21:53.28 | Charlie77 | ./configure |
| 21:53.34 | Charlie77 | make menuconfig |
| 21:53.41 | Micho123 | Charlie77, OK |
| 21:54.59 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:57.37 | Charlie77 | is it ok now? |
| 21:58.48 | Micho123 | Charlie77, yes the XXX disappear...In order to select a menu I should scroll to an entry and press enter? |
| 21:59.13 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 21:59.14 | Charlie77 | try space |
| 21:59.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:59.49 | Micho123 | Charlie77, space remove the * from an entry |
| 22:00.04 | Charlie77 | then press it again to have * there |
| 22:00.13 | Charlie77 | escape and save |
| 22:00.17 | Micho123 | Charlie77, Done...After that? |
| 22:00.22 | Micho123 | Charlie77, Done |
| 22:00.23 | Charlie77 | make && make install |
| 22:00.26 | Micho123 | Charlie77, then make? |
| 22:00.32 | Micho123 | Charlie77, OK |
| 22:01.08 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 22:02.18 | Charlie77 | is anyone willing to accept a job with custom AGI against a reasonable fee? I need to store fetched dtmf digits in a mysql database (meter readings) |
| 22:03.23 | Micho123 | Charlie77, the cdr_mysql still don't appear in asteriks folder |
| 22:04.43 | Charlie77 | did you install config files? |
| 22:05.10 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:05.38 | Micho123 | Charlie77, Which ones? |
| 22:06.12 | Charlie77 | you can install sample config files |
| 22:06.33 | Charlie77 | command is "make config" if i remember correctly |
| 22:06.45 | eppigy | negative |
| 22:06.52 | eppigy | make config installs boot scripts |
| 22:06.55 | eppigy | etc. |
| 22:06.57 | eppigy | make sample |
| 22:07.03 | Charlie77 | right :) |
| 22:07.04 | russellb | make samples |
| 22:07.05 | russellb | :) |
| 22:07.07 | eppigy | copies sample configs to your /etasterisk dir |
| 22:07.36 | eppigy | SMOKE PURP BY THE POUND |
| 22:07.53 | russellb | o.O |
| 22:08.00 | eppigy | 8[] |
| 22:08.06 | Micho123 | so make, make samples and make install |
| 22:08.18 | Charlie77 | you can make samples also later |
| 22:08.18 | eppigy | make install |
| 22:08.22 | eppigy | than make samples |
| 22:08.36 | Charlie77 | if you did make install, there is no need to do it again |
| 22:08.52 | eppigy | TRABAJO |
| 22:09.05 | Charlie77 | gentlemen, have a good time |
| 22:09.08 | Charlie77 | bye |
| 22:09.19 | *** part/#asterisk Charlie77 (n=tekach@89-212-30-165.dynamic.dsl.t-2.net) |
| 22:09.36 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 22:22.09 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 22:25.23 | korihor | hi :) how pause realtime agents? PauseQueueMember don't work for realtime agents :( |
| 22:31.18 | hardwire | would it be too difficult to just remove reailtime agents and store them as "paused" in ASTDB? |
| 22:31.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:40.08 | korihor | hardwire: thanks, I don't know that. Let me try |
| 22:43.18 | Micho123 | Hi all, I configured asterisk to save cdrs inside database...Everything looks fine except that asterisk server in not generating accounting code...this field still blank in the table cdr...How to configure asterisk to write a value per user inside this fileld? |
| 22:44.31 | rue_mohr | hah, for 3 days now I been setting autooffhook to 0 and wondering why it wasn't working |
| 22:44.57 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-757a7d80f3298645) |
| 22:45.23 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:46.28 | Micho123 | I guess I found it |
| 22:52.24 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 22:59.39 | *** join/#asterisk xorl (n=xorl@li30-130.members.linode.com) |
| 22:59.48 | xorl | Cannot find extension context 'SIP/858*******' |
| 23:00.13 | xorl | the sip context works for the physical phone line |
| 23:00.21 | xorl | why will it not work for another number |
| 23:00.38 | *** join/#asterisk telecos (n=sergio@34.167.219.87.dynamic.jazztel.es) |
| 23:15.37 | *** join/#asterisk EI5GTB (n=Paul@apollo.paulsnet.org) |
| 23:15.49 | EI5GTB | can someone telll me if 0.2 is the latest version od oslec? |
| 23:16.10 | saftsack | cvs/svn is the latest version |
| 23:17.23 | EI5GTB | i see.. |
| 23:17.41 | EI5GTB | theres only yht one svn source?> |
| 23:18.29 | saftsack | seems so |
| 23:19.08 | EI5GTB | http://svn.astfin.org/software/oslec/ |
| 23:19.09 | EI5GTB | ? |
| 23:20.10 | *** join/#asterisk SlicerDicer (n=SlicerDi@24-119-155-26.cpe.cableone.net) |
| 23:21.31 | *** join/#asterisk thansen (n=thansen@h-68-164-180-2.atlngahp.covad.net) |
| 23:28.35 | tzafrir_laptop | EI5GTB, yes |
| 23:28.45 | Defraz | does anyone have any experience with Cisco and dial-peers? |
| 23:29.17 | EI5GTB | tzafrir_laptop, thanks! |
| 23:33.13 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:35.40 | EmleyMoor | BT have a misprint in their tariff again! |
| 23:36.01 | EmleyMoor | (and I am trying to make a ratefile from it) |
| 23:36.14 | *** part/#asterisk xorl (n=xorl@li30-130.members.linode.com) |
| 23:38.43 | *** join/#asterisk andy-x_l (i=4009eef6@gateway/web/ajax/mibbit.com/x-d2abb30dc103025c) |
| 23:40.43 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 23:41.44 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:43.09 | *** join/#asterisk digitalirony (n=eric@nat/digium/x-b0affbca1ef79323) |
| 23:45.08 | *** join/#asterisk andy-x_l (i=4009eef6@gateway/web/ajax/mibbit.com/x-d2a9e6ece1e54e05) |
| 23:46.51 | rue_mohr | this is frustrating, these manuals for the ip601 seem to conflict like crazy |
| 23:48.29 | rue_mohr | who can tell me what the keypad icon is before I can find it in the manual(s) |
| 23:48.41 | CunningPike | Is voip-info kaput? |
| 23:48.59 | EmleyMoor | Looks like it |
| 23:49.53 | rue_mohr | looks like speed dial |
| 23:52.26 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:58.14 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |