15:59.43 | *** join/#asterisk infobot (i=ibot@rikers.org) |
15:59.43 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.4 (2009/08/11), 1.6.0.13 (2009/08/11), 1.4.26.1 (2009/08/11), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
15:59.48 | [TK]D-Fender | \o/ |
15:59.49 | [TK]D-Fender | TiMMEH! |
16:00.02 | [TK]D-Fender | ~infobot |
16:00.20 | brah | WUT!??! |
16:00.33 | rob0 | 'E's not dead, 'e's stunned! You stunned 'im! |
16:01.01 | [TK]D-Fender | goes to set his laser printers on "kill" |
16:01.22 | pugachevcobra | [TK]D-Fender: it seems something about challenge, the SIP debug for my softphones shows after the 1st unauthorized, a new register from the softphone with extra hashes... and the Nokia doesn't do that. Still, if it doesn't like being challenged, how can it register on an external provider? |
16:01.57 | lowtek | pugachevcobra: Maybe the external provider isn't challenging him enough... |
16:02.19 | lowtek | Nokia's these days ... |
16:02.40 | pugachevcobra | lowtek: I'm gonna check this right now, gonna capture the stream between the nokia and the provider |
16:04.48 | verywiseman | when it is should to use sip proxy with asterisk? it is better if there are some articles talk about that? |
16:05.34 | lowtek | verywiseman: It's a hardware sizing thing for the most part, 200 sip registrations asterisk will handle that and more just fine, but if you get into the thousands, something like OpenSER might be better. |
16:06.31 | verywiseman | lowtek, is that mean open ser work lonely or with asterisk? |
16:06.55 | bmoraca | wow |
16:07.00 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
16:07.33 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
16:09.41 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
16:10.05 | *** join/#asterisk rossand (n=rossand@bas1-ottawa11-1176120677.dsl.bell.ca) |
16:11.05 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
16:11.15 | ayeso | what is the /var/log/asterisk/master.db file? |
16:11.24 | pugachevcobra | [TK]D-Fender lowtek: http://pastebin.ca/1541681 the sniffed connection between nokia and EXTERNAL provider |
16:11.32 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
16:12.17 | pugachevcobra | sorry each entry is duplicated |
16:12.37 | *** join/#asterisk hfb (n=hfb@pool-98-112-210-252.lsanca.dsl-w.verizon.net) |
16:13.21 | *** join/#asterisk sjobeck (n=Adium@68.165.229.136) |
16:13.23 | raden_work | how can i forward calls from asterisk when people are out of office id like them to just be able to hit a button on there phone or dial the number they want forwarded too like *72,101 |
16:13.46 | *** part/#asterisk sjobeck (n=Adium@68.165.229.136) |
16:14.06 | telnettech | raden: that is normally a feature that is directly on the hardware device |
16:14.35 | brah | Like a "Follow me" function? |
16:14.50 | pugachevcobra | [TK]D-Fender , lowtek: as you can see, it challenged the nokia, and it answered properly |
16:15.17 | telnettech | raden: it is something that can be setup in your features.conf file |
16:15.24 | telnettech | thru asterisk though |
16:16.22 | *** join/#asterisk zeroHalo (n=zeroHalo@173.13.92.17) |
16:17.06 | brah | raden_work, http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe |
16:17.21 | pugachevcobra | ok, anyone else? |
16:18.44 | *** join/#asterisk garymc (n=GaryMck@host81-134-0-102.in-addr.btopenworld.com) |
16:19.10 | garymc | Hi anyone know the best RAID setup for an Asterisk box? especially as my main thing is call recording? |
16:20.00 | garymc | I can put 6 scsi drives in my server |
16:20.42 | *** join/#asterisk Chinorro (n=Chino@91.117.226.19) |
16:21.26 | *** join/#asterisk jeremib (n=Jeremi@75-147-226-84-Knoxville.hfc.comcastbusiness.net) |
16:21.57 | jeremib | I have the need to connect 4 FXO and 8 FXS channels to an asterisk box. What type of hardware should I be looking for? |
16:22.06 | rene- | garymc: i am thinking that a NAS solution would be nice for asterisk recording |
16:22.17 | rene- | never done it but surely must be cool |
16:24.20 | raden_work | is there like a GUI to handle extensions that each person could have a login for lets say 101 is leaving the office soo they could choose who there calls going to ? |
16:24.20 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-75.link.net.pk) |
16:25.02 | brah | Use follow me. |
16:26.14 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-75.link.net.pk) |
16:26.49 | *** join/#asterisk [T]ank (n=chwall@206.71.78.158) |
16:28.04 | [T]ank | has anyone ever experienced where xlite(free version) can make outbound calls just fine, but when an inbound call comes in it rings, and when I click on answer xlite shows that the call is established, but asterisk shows the call still ringing and the person calling is still hearing ringing and goes to voicemail. |
16:28.15 | [T]ank | xlite is not actually answering the call even though is looks like it is. |
16:29.03 | raden_work | brah, ok |
16:30.13 | brah | No need for a gui, just something like `exten => _50.,1,FollowMe(${EXTEN:2})` |
16:30.20 | leifmadsen | [T]ank: sounds like a firewall issue or something |
16:30.22 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
16:30.51 | leifmadsen | [T]ank: you need to do some 'sip debug' on the asterisk console -- I bet the INVITE is getting to the x-lite, but the 200 OK back from the x-lite is not happening |
16:36.10 | garymc | rene- what is a NAS solution? |
16:36.58 | [TK]D-Fender | [12:13]<raden_work>how can i forward calls from asterisk when people are out of office id like them to just be able to hit a button on there phone or dial the number they want forwarded too like *72,101 <-- its your dialplan, get coding. |
16:37.04 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
16:37.17 | [TK]D-Fender | raden_work: There aren't GUI's for tiny bits, only taking over everything typically. |
16:37.33 | [TK]D-Fender | And "app_followme = almost worthless |
16:37.44 | raden_work | [TK]D-Fender, i was just wondering people are gui happy around here dialing a phone to much work |
16:37.46 | brah | Almost is the keyword |
16:38.00 | raden_work | [TK]D-Fender, how would u set it up > |
16:38.09 | [TK]D-Fender | raden_work: This is simple dialplan |
16:38.10 | [T]ank | leifmadsen: mmmm.... could be firewall. unfortunatley i dont get to admin that. Crap... |
16:38.41 | [TK]D-Fender | ~book |
16:38.42 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
16:38.46 | [TK]D-Fender | ~infobot |
16:39.16 | raden_work | i have the book |
16:39.28 | [TK]D-Fender | raden_work: Just confirming the bot is fully back |
16:39.28 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
16:39.36 | [TK]D-Fender | raden_work: wasn't for you |
16:39.51 | beniwtv | Hi all... is there a way to get the SIP username on a SIP channel, originated from a softphone? |
16:40.21 | [TK]D-Fender | beniwtv: ${CHANNEL} |
16:40.44 | [TK]D-Fender | beniwtv: Or use SetVar in your peer definition to make something you don't need to aprse out |
16:41.00 | raden_work | oh ok |
16:42.11 | [TK]D-Fender | ~infobot |
16:42.12 | infobot | from memory, infobot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
16:42.16 | [TK]D-Fender | MY BITCH!!! |
16:42.19 | *** join/#asterisk mr_pete (n=Pete@80-195-172-76.cable.ubr01.sutt.blueyonder.co.uk) |
16:42.25 | [TK]D-Fender | ~areyouadog ? |
16:42.26 | infobot | Bark! Bark! |
16:42.29 | [TK]D-Fender | ~botsnack |
16:42.29 | infobot | [TK]D-Fender: aw, gee |
16:42.34 | [TK]D-Fender | infobot: Good boy! |
16:42.34 | infobot | [TK]D-Fender: :) |
16:43.01 | *** part/#asterisk [T]ank (n=chwall@206.71.78.158) |
16:44.05 | [TK]D-Fender | [12:19]<garymc>Hi anyone know the best RAID setup for an Asterisk box? especially as my main thing is call recording? <-- what defines "best RAID"? |
16:44.11 | [TK]D-Fender | garymc: What is your actual expected load? |
16:44.43 | brah | google a |
16:44.44 | [TK]D-Fender | garymc: Who shot JR? |
16:45.02 | garymc | well im expexting alot of calls in the long run |
16:45.03 | [TK]D-Fender | garymc: What is the average airspeed velocity of an unladen swallow? |
16:45.05 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
16:45.12 | garymc | hopefully a couple of hundred a day |
16:45.28 | [TK]D-Fender | garymc: SIMULTANEOUS. |
16:45.35 | garymc | yes |
16:45.56 | garymc | when things are going good anyway |
16:46.04 | [TK]D-Fender | garymc: Hundreds of simultaneous calls? |
16:46.08 | garymc | to start off , no where near that |
16:46.17 | garymc | nope 8 calls simultaneous |
16:46.26 | garymc | but a couple of hudnred a day |
16:46.34 | garymc | hopefully |
16:46.42 | [TK]D-Fender | garymc: 8 calls? thats in "who cares" territory. |
16:47.02 | [TK]D-Fender | garymc: FFS my watch can do that... and its ANALOG |
16:47.03 | garymc | well i only getting ISDN30e * channel to start off with |
16:47.15 | garymc | 8 channel |
16:47.24 | garymc | well if your watch can do it, i want one |
16:47.28 | garymc | :) |
16:48.17 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
16:48.26 | garymc | well if im recording a 100 calls a day im thinking it would use up alot of Disk space, so was wondering which RAID config would be best |
16:48.29 | [TK]D-Fender | garymc: Go ask Q for another.... |
16:48.42 | [TK]D-Fender | just wishes it would tell the TIME... |
16:48.49 | garymc | lol |
16:49.00 | garymc | im pretty sure Q is dead |
16:49.46 | garymc | [TK]D-Fender: Are you running Asterisk on a server, rack server? |
16:49.56 | *** join/#asterisk errotan (n=errotan@62.201.123.1) |
16:50.13 | garymc | [TK]D-Fender: I'll be running mine on a HP rack with 6 scsi bays |
16:50.23 | [TK]D-Fender | garymc: http://en.wikipedia.org/wiki/John_de_Lancie |
16:50.25 | garymc | so the RAID config will be pretty important |
16:50.26 | [TK]D-Fender | garymc: NOPE! |
16:50.47 | [TK]D-Fender | garymc: Of course my * is on a server. It SERVES people |
16:51.06 | mr_pete | has a rather n00b question if he may? |
16:51.25 | garymc | not sure noob questions go down too well here |
16:51.33 | Naikrovek | ask anyway |
16:51.37 | mr_pete | well everyone has to start somewhere :) |
16:51.49 | garymc | my sentiment exactly |
16:51.54 | manxpower | questions that can be easily answered by reading some docs don't go down well here, most others are OK |
16:51.57 | mr_pete | I've just migrated my SPA3102 from using Voxalot to using an internal asterisk box (on Unslung/Optware) |
16:52.06 | mr_pete | incoming/outgoing calls all behave |
16:52.17 | generalhan | MAN ... Q is OLD. that wiki pic does not do him the justice he deserves |
16:52.20 | mr_pete | but voicemail won't auto-id the user (always asks for mailbox num) |
16:52.22 | jeremib | how do I connect eithera block66 or rj11 connectors to a TDM2400P ? |
16:52.27 | garymc | Q is dead |
16:52.36 | [TK]D-Fender | mr_pete: Show us <- |
16:52.57 | manxpower | jeremib: The card has an AMP connecter. wire an amp cable into 66 block or rj11 |
16:53.05 | [TK]D-Fender | jeremib: its got an amphenol (RJ-21) on the back |
16:53.07 | garymc | ok, well Im wondering how the hell im gonna know which RAID config to use to setup an asterisk server? |
16:53.10 | mr_pete | callerid(num) to NoOp shows "Anonymous <anonymous>" |
16:53.21 | [TK]D-Fender | garymc: Well what are you considering? |
16:53.26 | mr_pete | what's the config paste URL address again? |
16:53.33 | beniwtv | [TK]D-Fender: thanks, that worked :) |
16:53.33 | [TK]D-Fender | mr_pete: What is the call coming from? |
16:53.39 | [TK]D-Fender | beniwtv: You're welcome |
16:53.45 | mr_pete | it's coming from the SPA3102, which has a user (1000) set |
16:53.48 | jeremib | thanks manxpower and [TK]D-Fender |
16:53.49 | manxpower | garymc: asterisk does not care anything about raid |
16:54.06 | mr_pete | and a displayname. It's also defined (I believe correctly) in sip.conf |
16:54.09 | [TK]D-Fender | garymc: And not knowing RAID means you don't know even server hardware basics. this has nothing to do with * |
16:54.10 | mr_pete | but I'm missing something.... |
16:54.12 | garymc | yes, but my disk drives that store call recordings etc do |
16:54.13 | mr_pete | (obviously) |
16:54.33 | garymc | yes ok |
16:54.35 | manxpower | garymc: no. It doesn't. Use whatever raid or non-raid you want to use. |
16:54.43 | pugachevcobra | [TK]D-Fender: you've seen the challenge? |
16:54.46 | [TK]D-Fender | mr_pete: pastebin the incoming call with SIP debug enabled, verbose 10, and include your sip.conf entry for it along with your DIALPLAN. |
16:54.47 | [TK]D-Fender | ~pb |
16:54.48 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
16:54.50 | [TK]D-Fender | ^^^^^^^6 |
16:55.07 | manxpower | infobot is back! |
16:55.16 | [TK]D-Fender | manxpower: Yeah, I followed up on it an hour ago |
16:55.24 | pugachevcobra | [TK]D-Fender: http://pastebin.ca/1541681 the sniffed connection between nokia and EXTERNAL provider... the provider challenges and the nokia answers |
16:56.15 | [TK]D-Fender | pugachevcobra: No idea... |
16:57.00 | pugachevcobra | [TK]D-Fender: ok, thanks anyway |
16:57.29 | mr_pete | have to see if I can trap this log first (crappy terminal) |
16:58.28 | *** join/#asterisk KrisWillis (n=kris@host86-159-73-243.range86-159.btcentralplus.com) |
16:59.20 | *** part/#asterisk jeremib (n=Jeremi@75-147-226-84-Knoxville.hfc.comcastbusiness.net) |
17:00.27 | KrisWillis | Good afternoon. Is it possible to display information on an IP phone screen based on which line a call has been received on? For example, if multiple companies calls are handled by the same group of people, they know which company the caller has dialed? |
17:01.07 | KrisWillis | I couldn't find any information on this in the O'Reilly book |
17:01.18 | generalhan | KrisWillis: i do this by setting the CALLERID(name)="$comapny_name. |
17:01.21 | [TK]D-Fender | KrisWillis: Unless the phone supports a proprietary header for indicating this, then all you can do is manipulate the Caller ID |
17:01.51 | [TK]D-Fender | KrisWillis: And the Book doesn't give specifics for much. Its for general theory and understanding. |
17:02.13 | KrisWillis | generalhan / [TK]D-Fender Thanks guys, I'll look into playing with the caller ID |
17:02.54 | *** join/#asterisk sjobeck (n=Adium@71-20-188-20.war.clearwire-wmx.net) |
17:03.10 | raden_work | we can switch ours todo diffrent rings based on inbound DID |
17:03.12 | *** part/#asterisk sjobeck (n=Adium@71-20-188-20.war.clearwire-wmx.net) |
17:04.03 | [TK]D-Fender | raden_work: Depends on your phone |
17:05.44 | raden_work | [TK]D-Fender, im googling and looking in book you said that "app_followme = almost worthless so what should i do ? |
17:05.56 | [TK]D-Fender | raden_work: Just code it yourself. |
17:06.13 | [TK]D-Fender | raden_work: "core show function DB" <- |
17:07.09 | mr_pete | try http://pastebin.ca/1541733 |
17:07.19 | raden_work | hmm |
17:07.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
17:07.25 | *** join/#asterisk jeremib (n=Jeremi@75-147-226-84-Knoxville.hfc.comcastbusiness.net) |
17:07.57 | jeremib | anyone have any good/back feedback on the astribank usb channel banks for asterisk by xorcom? |
17:08.26 | [TK]D-Fender | mr_pete: Where is your dialplan? What ver of *? |
17:08.40 | [TK]D-Fender | jeremib: What are your needs? |
17:08.52 | pugachevcobra | [TK]D-Fender: oh man i just found out |
17:08.55 | mr_pete | Asterisk 1.4.22.1 built by slug @ builder on a i686 running Linux on 2009-01-13 |
17:09.10 | [TK]D-Fender | mr_pete: DIALPLAN PLEASE |
17:09.19 | mr_pete | exten => *500,1,VoicemailMain(s${CALLERIDNUM}) |
17:09.19 | mr_pete | exten => *500,2,Hangup |
17:09.30 | mr_pete | that's all it has for that extension in [home] context |
17:09.33 | pugachevcobra | [TK]D-Fender: wrong realm, that's it... |
17:09.40 | [TK]D-Fender | pugachevcobra: SMRT :) |
17:09.48 | [TK]D-Fender | pugachevcobra: Glad you found it.... |
17:09.52 | pugachevcobra | [TK]D-Fender: weird though, as the softphones were using the wrong one but still connecting... |
17:10.02 | mr_pete | have tried ${CALLERID(all)} also |
17:10.03 | [TK]D-Fender | pugachevcobra: meh! |
17:10.06 | jeremib | Well, i'm replacing a traditional pbx system with an ip based one. There is 4fxo and 8fxs lines that need to be supported, the rest are viop phones. Trying to find a hardware solution for the fxo/fxs lines without a whole lot of knowledge |
17:10.12 | mr_pete | which is where I see the Anonymous output |
17:10.16 | [TK]D-Fender | mr_pete: neither will give you the NUMBER |
17:10.19 | pugachevcobra | [TK]D-Fender: thanks for the help anyways! |
17:10.27 | [TK]D-Fender | mr_pete: And you do not even set the callerid in your peer <0- |
17:10.47 | [TK]D-Fender | mr_pete: ${CAllERIDNUM} <- does not exist in 1.4 |
17:10.51 | raden_work | [TK]D-Fender, pardon my idiocy but im not seeing the relation between DB and what I want todo as far as how to use it |
17:10.56 | mr_pete | hence why I said I tried both :) |
17:11.03 | raden_work | and the asterisk book not much help either |
17:11.05 | pugachevcobra | cya all |
17:11.47 | [TK]D-Fender | raden_work: Make dialplan to ask the user where they want to forward calls to. STORE it in AstDB. In your extens that would dial the phone check if they HAVE a forwarding value stored in the DB and act ACCORDINGLY. |
17:12.20 | Kobaz | it seems that when i pass the 'r' option to a Queue, it doesn't play the periodic announcements... even though the config says it does |
17:12.21 | [TK]D-Fender | mr_pete: ${CALLERID(all)} <- and this will just report back what the SIP client says. You should be setting the CID in the PEER |
17:12.22 | *** join/#asterisk rossand (n=rossand@bas1-ottawa11-1176120677.dsl.bell.ca) |
17:12.29 | raden_work | ok |
17:12.33 | Kobaz | 'r' -- ring instead of playing MOH. Periodic Announcements are still made, if applicable. |
17:12.53 | [TK]D-Fender | blinks... |
17:12.56 | [TK]D-Fender | Kobaz: ... and? |
17:12.56 | manxpower | NO! |
17:12.58 | mr_pete | ok, that's a start to look at, ta. |
17:13.08 | Kobaz | [TK]D-Fender: it's a bug i assume? |
17:13.17 | [TK]D-Fender | Kobaz: I don't see you SHOWING ANYTHING |
17:13.23 | manxpower | "r" play ringing sound to caller even if the caller should hear other tones like busy, number out of service, your call cannot be completed, etc. |
17:13.44 | Kobaz | manxpower: which version? |
17:13.53 | manxpower | Kobaz: all versions of Asterisk |
17:13.55 | [TK]D-Fender | manxpower: ... context FAIL. go caffeinate :p |
17:14.12 | mr_pete | reload |
17:14.14 | Kobaz | manxpower: not on 1.6.0.10 |
17:14.17 | manxpower | you should (almost) never user "r" option to Dial |
17:14.43 | Kobaz | this isn;'t Dial, this is Queue |
17:15.09 | Kobaz | <PROTECTED> |
17:15.13 | Kobaz | that's all there is to show |
17:15.21 | Kobaz | it's not playing periodic annoucements |
17:15.22 | manxpower | Kobaz: that is for queues, totally different |
17:15.25 | [TK]D-Fender | Kobaz: PB the enire mess |
17:15.27 | Kobaz | manxpower: i know |
17:15.35 | Kobaz | [TK]D-Fender: well i can paste the the queue config, sec |
17:15.40 | [TK]D-Fender | manxpower: he was talking about queues from the start.... |
17:15.48 | manxpower | [TK]D-Fender: I realize that now. |
17:15.49 | [TK]D-Fender | manxpower: You. Caffeine. NOW :p |
17:15.53 | manxpower | [TK]D-Fender: I should. |
17:16.05 | manxpower | I have a better idea. |
17:16.07 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
17:16.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
17:16.32 | Kobaz | http://pastebin.ca/1541745 |
17:17.13 | mr_pete | [TK]D-Fender - got it working now, thanks :) |
17:17.51 | [TK]D-Fender | bokincrease your timeout on Queue() |
17:17.58 | [TK]D-Fender | Kobaz: increase your timeout on Queue() |
17:18.07 | Kobaz | which one |
17:18.11 | Kobaz | the call? |
17:18.12 | Kobaz | or the config |
17:19.09 | Kobaz | This particular queue is overloaded in function... There's a needed progression |
17:19.28 | Kobaz | Go to the bronze queue round robin, if noone picks up in 30 seconds, go to ringall |
17:19.42 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
17:19.44 | [TK]D-Fender | Kobaz: DO IT |
17:19.48 | Kobaz | so the timeout needs to be 30 seconds or less |
17:20.45 | Kobaz | okay i made the timeout 280 |
17:20.45 | Kobaz | <PROTECTED> |
17:20.46 | Kobaz | <PROTECTED> |
17:20.50 | Kobaz | no announcement |
17:21.06 | Kobaz | 75 seconds... no announcement |
17:21.23 | [TK]D-Fender | reaches for his ClueBat (tm)kobyou have no timeout for members in that queue |
17:21.35 | [TK]D-Fender | Kobaz: you have no timeout for members in that queue <-------- |
17:21.43 | Kobaz | oh yeah |
17:21.55 | [TK]D-Fender | FAIL |
17:21.55 | Kobaz | i forgot that queue's behave kinda stupidly with that |
17:21.59 | Kobaz | you need a timeout to play tracks, right? |
17:22.12 | [TK]D-Fender | Kobaz: Announcements don't play while the agent is RINGING |
17:22.22 | [TK]D-Fender | reaches for his ClueBat (tm) |
17:22.27 | Kobaz | oh yeah, that too |
17:22.35 | Kobaz | i need to rewrite the queue module :( |
17:22.43 | Kobaz | it's been driving me nuts for a year or more |
17:23.01 | Kobaz | tracks need to play while the phone is ringing |
17:23.08 | [TK]D-Fender | Kobaz: Stragedy <- Out-thinking yourself... |
17:23.19 | [TK]D-Fender | Kobaz: then you need a better plan. |
17:23.36 | Kobaz | i got around that by making my own musiconhold tracks, with a periodic announce pasted in the middle of songs |
17:23.49 | *** part/#asterisk jeremib (n=Jeremi@75-147-226-84-Knoxville.hfc.comcastbusiness.net) |
17:23.54 | [TK]D-Fender | Kobaz: WOW, and even a viable first guess! Hope for you yet! |
17:23.59 | Kobaz | heh |
17:24.11 | Kobaz | but it's still not the best way to do it |
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17:25.14 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
17:25.55 | [TK]D-Fender | Kobaz: Do you need to log who got the call? |
17:26.03 | Kobaz | yeah |
17:26.26 | [TK]D-Fender | Kobaz: DIAL, M() + m() could probably do the job |
17:26.46 | Kobaz | yeah |
17:27.18 | Kobaz | I still like having a queue though |
17:28.00 | Kobaz | i've been using AMI to get the call pickup infos |
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17:37.46 | generalhan | so, im still trying to get this folder creation for recorded queue calls taken care of, if anyone would like to take a look and drop some wisom on me!!! http://pastebin.com/d20aacc1e |
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17:46.27 | [TK]D-Fender | generalhan: You 100% sure the permissions are right on the folder? |
17:46.33 | [TK]D-Fender | genrI'd go check again... |
17:46.53 | generalhan | [TK]D-Fender: the issue is that there is no folder |
17:47.40 | generalhan | [TK]D-Fender: and the asterisk user as full rights to the parent folder |
17:47.53 | [TK]D-Fender | generalhan: Show.. |
17:48.09 | generalhan | [TK]D-Fender: this is a development only box. i am running * as root |
17:48.22 | generalhan | ..dont yell at me ! lol |
17:48.26 | [TK]D-Fender | generalhan: OK, I'll take that at face for now... |
17:48.33 | generalhan | lol |
17:48.48 | [TK]D-Fender | generalhan: That stupidity trumps permissions errors :p |
17:49.03 | generalhan | if i could just run more than one command with MONITOR_EXEC i would be all set, i figure |
17:49.13 | generalhan | [TK]D-Fender: stupidity trumps ALL |
17:50.35 | [TK]D-Fender | generalhan: IMO we need a better way to script what we want to do... |
17:51.01 | generalhan | i have this working on my production box, using local channels and a couple of macros to determine the rep that answered the call and creates the directory after i know that. but now im trying to use dynamic members for this queue, so i cant use the same strategy |
17:52.37 | generalhan | [TK]D-Fender: i dont really do many things that are too off-the-wall. for my needs * seems to allow me to do what i need it to, until this! |
17:54.45 | generalhan | ... maybe i just need a cig. lets try that. |
17:56.27 | [TK]D-Fender | generalhan: This would eb a minor patch, and a very deserving one. |
17:57.15 | [TK]D-Fender | generalhan: for the time being I might use the uniquid as the filename, and parse the queue log in a cleanup routine |
17:59.13 | generalhan | [TK]D-Fender: a patch to allow multiple commands with MONITOR_EXEC you mean ? |
17:59.47 | [TK]D-Fender | generalhan: Ata minimum to allow creation of the path |
18:00.25 | generalhan | [TK]D-Fender: i have never submitted any requests for anything before, or even bug reports. not too sure i even know how ! lol |
18:00.51 | [TK]D-Fender | generalhan: bugs.gigium.com |
18:00.55 | [TK]D-Fender | digium* |
18:01.17 | generalhan | would be nice to have like a -p option with the labeling to just create it like a mkdir |
18:01.45 | [TK]D-Fender | generalhan: as i mentioned, this should be a very tiny patch |
18:02.02 | generalhan | [TK]D-Fender: ok thanks for the advice i think i will go submit that request. since this is just a development machine maybe it would be good to go by the time i go production ! |
18:02.18 | [TK]D-Fender | generalhan: And IMO should be treated as a "bug" and get merged into whatever base branch we start with from 1.4 (IMO) |
18:02.34 | [TK]D-Fender | 1.2 cleary not. |
18:06.51 | raden_work | call forwarding seems soo complicated I thought there would be something in asterisk that made it simple that had alot of features |
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18:13.09 | [TK]D-Fender | raden_work: * is a toolbox, not a "model car in 5 pieces" |
18:13.31 | raden_work | its like learning a new language |
18:13.33 | [TK]D-Fender | raden_work: And this is simple. depending how lazy you want to be, under a dozen lines of dialplan. |
18:14.01 | [TK]D-Fender | raden_work: Your reaction to this gives an impression you've never programmed in your life. |
18:14.08 | raden_work | well i cant find examples to work off of thats how i understand things the best :( |
18:14.12 | [TK]D-Fender | radeit is a new language. that also isn't relevent. |
18:14.29 | raden_work | PHP, C, PERL, MYSQL |
18:14.38 | raden_work | thats where my experiance is |
18:14.52 | [TK]D-Fender | raden_work: GotoIf(), DB(), Read(), Set() <- basic building blocks that any programmer would pick up on almost instantly. |
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18:15.19 | raden_work | yeha i understand that |
18:15.28 | [TK]D-Fender | raden_work: Promtp for value, validate, store. check for value jump accordingly to DO X, Y, or Z based on the value |
18:15.36 | [TK]D-Fender | raden_work: So this is a non-issue |
18:15.58 | raden_work | but how do i set a variable from a phone to know they want it forwarded or there out of office |
18:16.00 | [TK]D-Fender | raden_work: actual 12 lines is more than you need at a minimum |
18:16.12 | [TK]D-Fender | raden_work: DB() <- like I told you from the start |
18:16.23 | [TK]D-Fender | raden_work: I did gift-wrap that answer for you. |
18:18.18 | raden_work | hmm ill have to keep reading |
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18:27.29 | timeshell_atwork | [TK]D-Fender Hi. Although I see a setting from CLI to set rx/tx gain for dahdi, I don't see a command to SHOW the current rx/tx gain in dahdi. Is there one? |
18:28.14 | kaldemar | dahdi show channel <channel> shows gains, iirc. |
18:28.22 | timeshell_atwork | Really? |
18:28.27 | timeshell_atwork | Let me check again |
18:28.47 | iCEBrkr | Qwell: You around? |
18:28.49 | generalhan | [TK]D-Fender: you think i should submit this "bug" under that Applications/app_mixmonitor category? |
18:29.02 | timeshell_atwork | kaldmar Nope... don't see gain |
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18:31.14 | raden_work | [TK]D-Fender, you wouldnt have an example script you could show me would u |
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18:33.53 | kaldemar | raden_work: is this what you're aiming for: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
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18:36.14 | kaldemar | that's some deprecated stuff, but it works as an example. if you didn't find that before, you really weren't looking. :P |
18:37.25 | ayeso | can someone shed some light on 'ulimit -l' settings? should I not change this? |
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18:51.15 | trebaum | I'm trying to get an E1 setup on a te400 series card, and i'm having some issues. The telecom said that the information I needed was in ITU-T I.431. I'm reading the tech spec and not getting much out of it. Anyone have an idea of what they are talking about? |
18:51.46 | trebaum | it's totally ok to tell me i'm stupid, but it's going to take me a lot longer to get this setup. |
18:53.19 | trebaum | I know that the signalling is euroisdn, though I'm trying to figure out the coding, framing, and timing. I have the jumpers setup on it for E1, and I keep getting a red alarm on the connection when running dahdi_scan |
19:00.28 | tzafrir_laptop | trebaum, just generate configuration with dahdi_genconf |
19:00.35 | tzafrir_laptop | it should be a good start |
19:00.57 | tzafrir_laptop | (ccs,hdb3) |
19:00.57 | *** join/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) |
19:01.53 | kb3ien | i'm on a budgetone 200, and all my trasfers are blind; this makes call parking rather unpleasant. Is there a way to do attended transfers? |
19:02.26 | voipmonk | ive got some thermite for your barbietone |
19:03.08 | Da-Geek | Hi All, Looking for a VoIP/SIP provider/Gateway (In the UK) that can rival/beat BT Business Plans... Anyone got any that would like to suggest ? |
19:03.41 | Da-Geek | voipmonk: im not sure thermite would do the jom on a Budgetone ;-) |
19:03.48 | Da-Geek | *job |
19:04.28 | Qwell | iCEBrkr: nope |
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19:04.41 | iCEBrkr | Qwell: Well get over here :P |
19:04.55 | Qwell | who what? |
19:05.03 | Qwell | no, I haven't had a chance. :( |
19:05.20 | iCEBrkr | feh, like you do work :P |
19:05.38 | Qwell | I do *some* |
19:06.06 | Qwell | iCEBrkr: where were you this weekend when I wiped my phone before running a backup? :P |
19:06.21 | Qwell | could've told me there was restore functionality now :D |
19:06.30 | iCEBrkr | Qwell: I saw you had some comments in the bug tracker about someone unable to cross-compile for ARM... Meanwhile it wasn't a bug. But nayhow.. |
19:06.49 | Qwell | wasn't a bug? |
19:06.50 | iCEBrkr | Qwell: You got any references I can use for cross-compiling astrisk? :P |
19:07.01 | Qwell | ./configure --host= |
19:07.02 | Qwell | basically |
19:07.03 | iCEBrkr | Qwell: Yeah, in the end the guy just didn't know what he was doing :) |
19:07.08 | Qwell | err, --target? whichever |
19:07.13 | iCEBrkr | orly? |
19:07.23 | iCEBrkr | I thought there was more environment stuff to setup |
19:08.02 | iCEBrkr | Qwell: uhh.. This weekend, I was wasted.. Ala Damin/Kris style. |
19:08.15 | iCEBrkr | I just didn't have a piano to passout under. |
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19:24.20 | trebaum | tzafrir_laptop: Thanks for the dahdi_genconf tidbit. That was helpful. Though i'm still getting a red alarm on span 1... doesn't a red alarm mean that there is something wrong with the configs? |
19:25.00 | tzafrir_laptop | either that, or simply no cable connected |
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19:25.33 | trebaum | I thought about the cable thing... but here is the issue. I'm doing all of this remotely, and I don't have physical access to the box. |
19:25.57 | trebaum | So I have to rely on the tech that is onsite to tell me that the cable is plugged in. He says it is. |
19:26.19 | trebaum | So if we take that out of the equation for now, what does that leave? |
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19:27.07 | tzafrir_laptop | head -n 3 /proc/dahdi/1 |
19:27.12 | rickross | hi Qwell - you closed this ticket as "fixed - https://issues.asterisk.org/view.php?id=13375 - I believe I just experienced the exact problem using 1.6.1.1 built today |
19:27.29 | trebaum | output: Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RED |
19:27.36 | trebaum | output: 1 TE4/0/1/1 Clear (In use) RED(SWEC: MG2) |
19:32.42 | Qwell | rickross: Then you need to install sounds that have been installed by default for nearly 18 months. |
19:33.20 | rickross | hhmm, I AM using a 3rd-party sound package |
19:33.39 | rickross | there's no CLI output of an attempt to play a missing sound |
19:33.57 | Qwell | If you aren't seeing the same messages, it isn't the same issue. |
19:34.41 | rickross | when there is no temp greeting in the user's voicemail dir, then the "press 0" option behaves normally, and all prompts seem to play normally |
19:35.13 | rickross | if I record a temp greeting, then access the voice mailbox again, pressing "0" for more options causes immediate exit |
19:35.14 | Qwell | Does the prompt in his warning exist? |
19:36.07 | rickross | vm-tmpexists ?? |
19:36.11 | rickross | I will check now |
19:36.57 | *** part/#asterisk zeroHalo (n=zeroHalo@173.13.92.17) |
19:37.01 | raden_work | exten => _*21X.,5,SayDigits(${EXTEN:3}) |
19:37.01 | raden_work | <PROTECTED> |
19:37.32 | rickross | no, there is no vm-tmpexists in my sounds dir |
19:37.54 | grandpapadot | raden_work: reads X. (subtracts first 3 digits) |
19:38.02 | Qwell | rickross: well then... |
19:38.10 | raden_work | grandpapadot, thanks :) |
19:38.18 | grandpapadot | raden_work: *21805 would return 805 |
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19:38.54 | rickross | Qwell, why wouldn't my CLI show any kind of log message that it attempted to play a soundfile which does not exist? |
19:39.09 | lapietra | Hi to all... |
19:39.26 | Qwell | rickross: Because FreePBX defaults to turning useful logging off. |
19:39.32 | Qwell | Go ask them. |
19:39.33 | rickross | LOL |
19:39.50 | lapietra | Hi can somebody Help me... |
19:40.24 | lapietra | y have a problem whit my dahdi 2.2 and asterisk 1.6.0 whit a te220 p |
19:40.30 | grandpapadot | lapietra: you don't have to ask for permission to ask for help, just state your problem/question/concern and deposit $0.25 in the slot. |
19:41.01 | lapietra | Ok... |
19:41.15 | lapietra | i somebody can Helpme i can deposit it |
19:41.18 | lapietra | XD |
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19:47.34 | raden_work | [Aug 24 14:44:49] WARNING[723]: pbx.c:3080 pbx_extension_helper: No application 'DBput' for extension (to-callcentric, *21102, 2) |
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19:49.49 | kaldemar | raden_work: function DB() has replaced app DBput |
19:50.11 | raden_work | ahh sorry |
19:50.28 | raden_work | im soo sick of everything changing :( |
19:50.37 | lapietra | [Aug 24 15:50:23] WARNING[8291]: chan_dahdi.c:11612 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
19:52.01 | grandpapadot | raden_work: Use the version that works for you, we have no plans to upgrade from 1.4 to 1.6 |
19:52.05 | kaldemar | raden_work: it's been 4 years since DB() got introduced. |
19:52.18 | raden_work | well asterisk book is a lil behind then |
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19:53.56 | kaldemar | the 2nd edition is written for 1.4. |
19:55.06 | raden_work | [Aug 24 14:53:01] WARNING[755]: pbx.c:3080 pbx_extension_helper: No application 'DB' for extension (to-callcentric, *21102, 2) |
19:55.06 | raden_work | <PROTECTED> |
19:55.17 | raden_work | im running 1.6 :( |
19:55.27 | kaldemar | like said earlier, DB is not an application, but a function. |
19:55.38 | grandpapadot | raden_work: There is no DB application, use Set |
19:55.50 | raden_work | exten => _*21X.,n,DB(CFIM/${CALLERIDNUM}=${EXTEN:3}) |
19:56.03 | raden_work | so set instead of DB ? |
19:56.12 | kaldemar | so you don't use it like exten => 123,1,DB() but exten => 123,1,Set(DB(foo/bar)=111) |
19:57.04 | raden_work | this is why i have soo many issues and seem like such a idiot people send me files that work then they dont i look at book make changes that dont work google other stuff dont work now your showing me something so completly diffrent again |
19:57.18 | kaldemar | and the book does have examples on DB |
19:57.59 | kaldemar | pages 160- |
19:59.18 | raden_work | exten => _*21X.,n,SET(DB(CFIM/${CALLERIDNUM}=${EXTEN:3})) |
19:59.18 | raden_work | exten => _*21X.,n,SayDigits(${EXTEN:3}) |
19:59.24 | raden_work | am i doing something wrong ? |
19:59.32 | [TK]D-Fender | raden_work: Yes. |
19:59.37 | raden_work | :( |
19:59.44 | seanbright | exten => _*21X.,n,Set(DB(CFIM/${CALLERIDNUM})=${EXTEN:3}) |
19:59.50 | [TK]D-Fender | raden_work: one of those vars no longer exists |
20:00.21 | [TK]D-Fender | raden_work: And the brackets. And I'm worndering if you actually have a "1" priority there somewhere |
20:00.36 | raden_work | yes at answer |
20:00.41 | raden_work | it reads back extension |
20:00.41 | fun330 | what is the best way to handle a multi site phone system? should i use point to point t's or just the interenet? also should i use a colo for the main system? |
20:00.42 | [TK]D-Fender | raden_work: And I can see yuo are desperately trying to copy&paste other peoples outdated crap samples |
20:00.44 | raden_work | just doesnt forward |
20:00.45 | Katty | stretches. |
20:00.59 | [TK]D-Fender | fun330: Depends |
20:01.08 | Katty | hello all you beautiful people! |
20:01.10 | rob0 | My number one priority is to have number one priorities for all extensions. |
20:01.16 | voipmonk | heh |
20:01.21 | Katty | my priority is to hug rob0 |
20:01.23 | [TK]D-Fender | raden_work: Naturally it doesn't. you're just invented a DB value. It has no functioanlity attached to it |
20:01.30 | rob0 | hugs katty |
20:01.37 | Katty | :> |
20:01.39 | Katty | hugs rob0 |
20:01.40 | [TK]D-Fender | raden_work: You modify the EXTESIONS you use to dial your phones to LOOK for those values |
20:02.02 | raden_work | im not getting yeah |
20:03.03 | Katty | i feel like watching all the 1970 sesame street movies. |
20:03.15 | jaytee | hugs Katty |
20:03.26 | jaytee | Manamana |
20:03.28 | Katty | :>>> |
20:03.31 | Katty | do do be do do |
20:03.36 | [TK]D-Fender | Katty: Mew. |
20:03.36 | Katty | hugs jaytee |
20:03.38 | Katty | [TK]D-Fender: hello. |
20:03.43 | jaytee | do do be do |
20:04.15 | eppigy | TRABAJO |
20:04.28 | Katty | do do be do do |
20:04.30 | Katty | be do do |
20:04.32 | rob0 | Hi ho, Kermit The Frog, here, live for Sesame Street News. Little Jack Horner, why are you sitting there? |
20:04.35 | Katty | be do be do be do do do do do! |
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20:04.40 | voipmonk | talaga? |
20:05.02 | Katty | infobot: mahnahmahnah? |
20:05.08 | Katty | infobot: mahna mahna? |
20:05.27 | raden_work | [TK]D-Fender, im not quiete understanding you |
20:05.27 | Katty | infobot: mahna mahna is http://www.youtube.com/watch?v=hTkGXuiT55w |
20:05.27 | infobot | Katty: okay |
20:05.42 | [TK]D-Fender | raden_work: What part? |
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20:06.12 | Katty | infobot: forget mahna mahna |
20:06.13 | infobot | Katty: i forgot mahna mahna |
20:06.30 | Katty | infobot: mahna mahna is http://www.youtube.com/watch?v=ynjIoymWHvU |
20:06.31 | infobot | Katty: okay |
20:06.45 | Qwell | ~roflmao? |
20:06.46 | infobot | hmm... roflmao is rolling on the floor laughing my arse off, or painful, or http://www.youtube.com/watch?v=iEWgs6YQR9A |
20:07.03 | Katty | Qwell: yesh, that too. |
20:07.17 | Katty | infobot: porn |
20:07.17 | infobot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
20:07.23 | Katty | hmm. |
20:07.33 | Katty | we must get that other wow song in there, somehow. |
20:07.50 | rickross | Qwell, it was definitely the missing sound prompt - although I had full logging turned on, and Asterisk was giving no indication re: the missing file - simply dumped |
20:07.55 | Katty | infobot: wow? |
20:07.59 | rickross | http://pastebin.ca/1541934 |
20:08.00 | Katty | infobot: world of warcraft? |
20:08.01 | infobot | rumour has it, world of warcraft is for porn |
20:08.07 | Katty | :>>>>>>>>>>>>> |
20:08.17 | rickross | anyway, thanks for the insight - problem solved |
20:09.10 | Katty | infobot: ode to joy? |
20:09.40 | [TK]D-Fender | ~habanera |
20:09.41 | infobot | well, habanera is bork Bork BORK Bork! http://www.youtube.com/watch?v=EDFgtFXfnv0 |
20:10.00 | Katty | i wonder how infobot forgot ode to joy :< |
20:10.17 | [TK]D-Fender | uurrr de de de doooo de do BORK BORK BORK! |
20:10.56 | *** join/#asterisk Emrah86 (n=asterisk@a37254.upc-a.chello.nl) |
20:11.01 | Katty | infobot: Ode to Joy is meep! meep! meep! http://www.youtube.com/watch?v=xpcUxwpOQ_A |
20:11.02 | infobot | Katty: okay |
20:11.14 | Katty | infobot: forget Ode to Joy |
20:11.15 | infobot | i forgot ode to joy, Katty |
20:11.48 | Katty | is it Me or Mi? |
20:12.05 | Katty | that beaker does. |
20:12.09 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
20:12.42 | Katty | infobot: Ode to Joy is mi! mi! mi! http://www.youtube.com/watch?v=xpcUxwpOQ_A |
20:12.42 | infobot | Katty: okay |
20:13.42 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
20:13.57 | Emrah86 | hi all... do you know a gui for CDR's for customers? the idea is that a customer can check his own called numbers. I saw something like asterisk2billing, but it didnt work for me... any ideas would be great... |
20:14.53 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
20:15.10 | Katty | webstat |
20:15.12 | [TK]D-Fender | Emrah86: asteriskstat |
20:15.15 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
20:15.32 | Katty | ah right. that one |
20:15.34 | Katty | not webstat |
20:16.10 | Katty | eppigy: help me pick out a recipe for tomorrow |
20:16.18 | Emrah86 | [TK]D-Fender, but that one isnt secure... they can see everything.. i just want to build a web gui for it where users log in, and only see their own cdr's |
20:16.26 | *** join/#asterisk AlexTO (n=aacm_ale@190.26.169.215) |
20:16.35 | AlexTO | Hi Everyone |
20:16.39 | Emrah86 | hi AlexTO |
20:16.52 | AlexTO | can someone give me a hand with a TDM410 |
20:17.13 | Katty | eppigy: haven't made the banana pudding yet :< not had time. |
20:17.43 | Emrah86 | ahh banana pudding !! damn thank you for your reminder, i have one in the fridge! |
20:20.08 | [TK]D-Fender | Emrah86: Not too many 3rd party with security . usually those that do have en entire framework around them. |
20:20.31 | [TK]D-Fender | Emrah86: You could do a DB trigger or similar to restrict. |
20:21.46 | Emrah86 | [TK]D-Fender hmm i could also just use the framework of asterisk-stat and build a security check in front... |
20:21.52 | Emrah86 | hmm oke oke :) thats great, thnx :) |
20:22.14 | Katty | finds recipe for egg salad sandwiches |
20:22.26 | raden_work | how is one suppose to know how to work with asterisk is there a user manual for 1.6 ? |
20:22.32 | Katty | what goes well with egg salad sandwiches? |
20:22.35 | *** join/#asterisk pugachevcobra (n=fighter@189.107.7.26) |
20:22.37 | raden_work | all information i find is diffrent this is very confussing |
20:22.43 | Katty | infobot: the book? |
20:22.47 | Katty | infobot: thebook? |
20:22.48 | infobot | from memory, thebook is 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://astbook.asteriskdocs.org |
20:22.55 | raden_work | i have da book |
20:23.03 | Katty | ah, i thought the book was 1.6 |
20:23.05 | pugachevcobra | Hi all, my asterisk 1.6 is not logging CDR's, can anyone help on that? |
20:23.15 | [TK]D-Fender | raden_work: there are wonderful CHANGES and README files in your source tarball, etc |
20:23.16 | raden_work | Katty, its 1.4 :( |
20:23.36 | [TK]D-Fender | raden_work: And docs that show you how to use channel variables, which are standard, etc... |
20:23.45 | raden_work | Katty, tuna and cheese macaroni salad |
20:23.49 | [TK]D-Fender | raden_work: And most of 1.4 works in 1.6 |
20:23.55 | raden_work | 'most' |
20:24.07 | [TK]D-Fender | raden_work: some stuff in the book is still 1.2 however which is very unfortunate |
20:24.17 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
20:24.45 | jaytee | asteriskstat looks like it hasn't been maintained since 2005 |
20:24.48 | [TK]D-Fender | installing his new * server :D |
20:25.01 | [TK]D-Fender | jaytee: Yes, well * CDR hasn't changed since then :p |
20:25.20 | Skeeter- | amportal wont start on ubuntu 9.04 |
20:25.22 | kaldemar | raden_work: UPGRADE.txt in 1.6.0 and UPGRADE-1.6.txt in 1.6.1 source packages tells you that sort of changes from 1.4 to 1.6. |
20:25.45 | [TK]D-Fender | jaytee: Dell PE R610 Quad-core, 4 Gig, SAS RAID 5 |
20:25.52 | [TK]D-Fender | Skeeter-: GUI's are NOT supported here |
20:26.06 | [TK]D-Fender | ~freepbx |
20:26.07 | infobot | [~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 |
20:26.08 | Katty | jaytee: it still works nicely :P |
20:26.17 | jaytee | Katty, on 1.4? |
20:26.39 | Katty | yes. we've not gone to 1.6 yet due to isymphony not having 1.6 support yet. |
20:26.58 | Katty | infobot: seen seanmh? |
20:26.59 | infobot | seanmh is currently on #asterisk, last said: 'I'll let you know when we have 1.6 support and then I will await my cookie :D'. |
20:26.59 | generalhan | [TK]D-Fender: when youre all done, let me know how you like that server. i went with the 410 to save some cash. |
20:27.06 | seanmh | here here.. |
20:27.10 | seanmh | working on 1.6 support |
20:27.14 | [TK]D-Fender | What is it with companies shoving a #&^$ing "i" in front of everything? |
20:27.18 | jaytee | I'll have to play with it some on my failover server to see how it plays |
20:27.20 | seanmh | trying to get a beta out with 1.6 support within a week or |
20:27.21 | seanmh | so |
20:27.22 | Katty | no cookie yet :< |
20:27.27 | Katty | yay :> |
20:27.29 | seanmh | bear with us.. it's a lot of work |
20:27.34 | seanmh | :D |
20:27.36 | [TK]D-Fender | generalhan: I have PC towers much louder than this 1U. it's so f'n quiet its scary... |
20:27.43 | Katty | seanmh: i understand. |
20:27.43 | rob0 | [TK]D-Fender: iDunno ? |
20:27.48 | Katty | seanmh: we are very patient. |
20:27.54 | seanmh | heh.. yes you are |
20:27.59 | Katty | dangles cookie |
20:28.14 | generalhan | [TK]D-Fender: hmmm, i shoulda spent the extra money then. this one is LOUD. and periodically gets louder and softer, like the fans are playing music for me :( |
20:28.27 | [TK]D-Fender | rob0: There is no "i" in TEAM, but there is a "U" in DUMBASS :p |
20:28.38 | generalhan | haah |
20:28.38 | jaytee | lol |
20:28.44 | Katty | rob0: i'd thwap him. |
20:28.48 | [TK]D-Fender | generalhan: I can whisper over it... |
20:28.50 | Katty | thwaps [TK]D-Fender |
20:28.57 | generalhan | ok i need food |
20:28.59 | generalhan | brb |
20:29.59 | Katty | baby ducks go peepeepeep :> |
20:30.04 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
20:30.21 | rob0 | Thwank you Katty for the thwap. |
20:31.05 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/x-jowgeuvzboenhnar) |
20:31.32 | yziquel | hi. how can i check which codec is used during a given call? |
20:31.39 | *** join/#asterisk fskrotzki (n=fskrotzk@mail.perspectivepartners.com) |
20:31.40 | *** join/#asterisk Naikrovek (n=jeremiah@98.214.112.102) |
20:31.43 | [TK]D-Fender | generalhan: Holy McFuck this thing is fast... I didn't have enough time to exit my SSH connection when issuing a reboot :) |
20:31.55 | [TK]D-Fender | generalhan: before it cut me off :) |
20:32.05 | [TK]D-Fender | yziquel: "sip show channels" |
20:32.07 | lapietra | core show translation |
20:32.08 | lapietra | <PROTECTED> |
20:32.10 | lapietra | <PROTECTED> |
20:32.12 | lapietra | <PROTECTED> |
20:32.14 | lapietra | <PROTECTED> |
20:32.16 | lapietra | <PROTECTED> |
20:32.17 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
20:32.18 | lapietra | <PROTECTED> |
20:32.19 | lapietra | <PROTECTED> |
20:32.21 | lapietra | <PROTECTED> |
20:32.22 | *** kick/#asterisk [lapietra!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
20:32.31 | [TK]D-Fender | WRONG ANSWER |
20:32.42 | Naikrovek | you love that |
20:32.42 | Katty | http://www.youtube.com/watch?v=ficwZQYmRLE <- eppigy |
20:33.22 | yziquel | thanks. |
20:33.44 | *** join/#asterisk af_ (n=getsmart@88-149-240-57.dynamic.ngi.it) |
20:33.49 | Katty | ducky got schnozzled upon! |
20:34.25 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
20:34.39 | *** join/#asterisk R0dya (n=R0dya@78.16.213.190) |
20:36.53 | *** join/#asterisk jkroon (n=jkroon@dsl-240-155-226.telkomadsl.co.za) |
20:37.52 | [TK]D-Fender | jaytee: Oh, and I "Mr. Coffee"'d my IP 600 earlier today... still going strong :) |
20:37.58 | [TK]D-Fender | jaytee: Even the speakerphone :) |
20:38.19 | jaytee | hahaha |
20:38.49 | [TK]D-Fender | Server updated! |
20:38.51 | [TK]D-Fender | \o/ |
20:38.55 | jaytee | yay |
20:39.11 | [TK]D-Fender | yay, tomorrow is hopefully the big install prep day |
20:39.13 | jaytee | Katty did you install asteriskstat? |
20:39.20 | [TK]D-Fender | ok, I'm off home, BBIAB |
20:39.25 | jaytee | later |
20:39.50 | [TK]D-Fender | if lapietra shows up just remind him not to spam... |
20:40.15 | yziquel | and how can how check which codecs are available from my voip provider. it seems that i'm converting LAN gsm to outbound alaw, which seems pointless to me. |
20:40.32 | grandpapadot | yziquel: ask them |
20:41.13 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
20:41.18 | yziquel | grandpapadot: unfortunately there not very responsive. can you read that from the SIP/SDP requests? |
20:41.49 | grandpapadot | yziquel: find a better itsp ... |
20:41.52 | grandpapadot | ~itsp |
20:41.53 | infobot | [~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 |
20:42.23 | Katty | jaytee: yesh. |
20:42.51 | shadowalke | Hello i am currently running asterisk 1.6.1 and setting up sip.conf for realtime use for some reason when adding a user to the mysql DB it is not detected in asterisk |
20:43.42 | jaytee | Katty what distro are you using? |
20:44.18 | yziquel | grandpapadot: that's something i agree on. but it doesn't solve my immediate problem. |
20:47.32 | *** part/#asterisk asterwiki (n=asterwik@69.77.169.14) |
20:48.02 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
20:48.46 | bmoraca | yziquel: very few ITSPs support anything other than alaw/ulaw or g729, and the only way you'll know is if you ask them. |
20:48.56 | bmoraca | or try |
20:50.46 | *** join/#asterisk R0dya (n=R0dya@78.16.213.190) |
20:51.11 | *** join/#asterisk Naikrovek (n=jeremiah@98.214.112.102) |
20:55.16 | *** join/#asterisk rossand (n=rossand@bas1-ottawa11-1176120798.dsl.bell.ca) |
21:01.34 | jkroon | hi, the t1e1override option for wct4xxp (Digium, Inc. Wildcard TE205P dual-span T1/E1/J1 card 5.0V), does that serve a purpose wrt the hardware, or is T1/E1 determined by some chip on the board? |
21:03.00 | kaldemar | jkroon: you can set it with that module parameter |
21:03.15 | voipmonk | yep |
21:03.24 | jkroon | seeing that I'm using E1 and dahdi_genconf kept insisting it's T1 I set the t1e1override option to 0xFF and now the genconf generates a valid config but I still get no sync. |
21:03.44 | *** join/#asterisk propellerhead (n=yogurt2u@190.246.145.56) |
21:03.45 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:03.48 | jkroon | I plug the cable into a CISCO 2600 and it syncs immediately. so the PRI link from the telco seems to be fine. |
21:04.14 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
21:04.24 | bmoraca | jkroon: make sure your cable is wired properly. you may need to roll the T1 or use a crossover |
21:05.15 | jkroon | bmoraca, the cable works going into a CISCO 2600 ... surely that would indicate the cable is ok? |
21:05.21 | jkroon | or am i way off track? |
21:05.45 | bmoraca | not if the port in the 2600 is wired differently from the digium card |
21:06.56 | jkroon | interesting ... they're both using RJ45. You're telling me there is different standards? |
21:07.43 | bmoraca | technically, they're RJ48C, but, yes, just because the jack accepts 8P8C plugs does not mean they're internally wired the same |
21:08.20 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
21:08.43 | bmoraca | case in point: Adtran TA900s. The DSX ports on the 900e series require you to use a T1 crossover cable when connecting to customer hardware, where with the DSX ports of the 900 series you can use a straight-through |
21:09.22 | jkroon | mutters something not so nice about the telecommunications industry. |
21:09.30 | jkroon | ok, where do I locate the specs? |
21:09.53 | bmoraca | jkroon: which you need just depends on how the telco wired the jack at the d-marc |
21:10.00 | rob0 | My spex are right here on my nose! |
21:10.06 | voipmonk | :) |
21:10.33 | bmoraca | if you're plugging straight in to the smartjack, then you should only need a straight-through cable, and you're dealing with configuration. if it's been extended, it depends |
21:10.38 | jkroon | bmoraca, well, I know it works with the 2600, so if I can get it's pin-out, and I can get the Digium pinout I should be able to deduce what needs to go where and build the appropriate cable :). |
21:10.39 | bmoraca | s/d-marc/demarc |
21:10.59 | *** join/#asterisk galeras (n=galeras@186.80.181.115) |
21:11.04 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
21:11.12 | bmoraca | jkroon: T1 crossovers are easy enough to make...why not try it and see if it works? |
21:11.26 | jkroon | ok, for that matter - any recommendations on good reading material re ISDN concepts? |
21:11.44 | jkroon | starting with basics preferably. |
21:11.46 | bmoraca | jkroon: swap pins 1 with 4 and 2 with 5 and bam, you have a T1 cross |
21:12.20 | jkroon | bmoraca, guessing that's worth a try. |
21:12.27 | kaldemar | jkroon: http://kb.digium.com/entry/17/ |
21:12.52 | kaldemar | the knowledge base is pretty useful with that kinds of things. |
21:14.03 | jkroon | kaldemar, i only seem to see the question ... not the explanation? |
21:14.14 | yziquel | any free software softphone implementing g723.1? |
21:17.25 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
21:17.32 | *** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr) |
21:19.35 | yziquel | i believe the g723.1 has expired... so perhaps... |
21:19.43 | yziquel | the g723.1 patent |
21:21.30 | *** join/#asterisk DaveWelsh (n=dave@ottawa-hs-69-20-226-218.s-ip.magma.ca) |
21:22.27 | DaveWelsh | I have Polycom IP550 phones. Can anyone suggest a wired headset? I don't want wireless ones because I don't want people wandering around. |
21:23.04 | [TK]D-Fender | DaveWelsh: Plantronics M22 Amp + H261n binaural headset on Polaris quick-connect |
21:27.25 | grandpapadot | lol @ DaveWelsh |
21:27.36 | grandpapadot | ... tethering via polycom |
21:28.17 | *** join/#asterisk luminforce (n=luminbla@cpe-70-112-179-169.austin.res.rr.com) |
21:29.18 | luminforce | is there a way to add an entire subnet as a single entry/peer in sip.conf? |
21:29.23 | DaveWelsh | Thanks [TK]D-Fender. I'll check that out. |
21:29.52 | [TK]D-Fender | luminforce: no, thats what permit/deny is for |
21:30.06 | [TK]D-Fender | luminforce: host=dynamic in that case |
21:33.00 | luminforce | so... if i wanted to always send to one gateway, but receive from an entire subnet... could i set up one peer with the host=gateway.ip and another with host=dynamic permit=ip/mask? |
21:33.03 | shadowalke | Good afternoon everyone |
21:33.16 | shadowalke | could someone give me some insight as to why my sip.conf file is not working from realtime |
21:33.21 | shadowalke | I have extensions working just fine |
21:33.46 | *** join/#asterisk dparker (n=dparker@69.60.123.206) |
21:36.15 | leifmadsen | luminforce: not quite -- the host=dynamic means the peer has to register, which will still only let you receive from the registered peer. Just setup a peer that will be authenticated by username and password, then use permit and deny to control the access by IP, and you'll probably need insecure=invite,port |
21:40.41 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.77.124) |
21:40.45 | kb3ien | what is the best way to do a case-insensitive string comparison. seems that shell isn't avail in my make menuconfig. |
21:40.46 | DelphiWorld | hi |
21:40.58 | kb3ien | what version has func shell anyway? |
21:41.03 | *** join/#asterisk korihor (n=korihor@190.205.251.61) |
21:41.03 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
21:44.24 | DelphiWorld | asterisk do jingle? |
21:44.31 | [TK]D-Fender | kb3ien: AGI |
21:45.04 | [TK]D-Fender | DelphiWorld: http://www.google.ca/#hl=en&source=hp&q=asterisk+jingle&btnG=Google+Search&meta=&aq=f&fp=a1047c2a76fad57b |
21:45.51 | jkroon | any guesses as to why (using the exact same dialplan and call sequence as per comparison from CLI output) on my E1 link (in China) would some outbound calls correctly present CLI whilst others won't? |
21:46.03 | shadowalke | am i visable? |
21:46.33 | [TK]D-Fender | shadowalke: Yes, now bag it before you scare someone :p |
21:46.52 | jkroon | http://kb.digium.com/entry/23/ <-- needs updating, the te210/205p cards are now also handled by wct4xxp and no longer wct2xxp. in case someone here has edit rights. |
21:47.04 | [TK]D-Fender | shadowalke: " host=gateway.ip and another with host=dynamic permit=ip/mask? <- No. Host = must come from there. |
21:47.55 | [TK]D-Fender | jkWheres the comparative debug? |
21:47.57 | kb3ien | what is the best way to do a case-insensitive string comparison? sans agi? |
21:48.05 | [TK]D-Fender | jkroon: Wheres the comparative debug? |
21:48.18 | DelphiWorld | [TK]D-Fender: got it...voipuinfo |
21:48.28 | [TK]D-Fender | kb3ien: Perhaps REGEX() |
21:48.47 | kb3ien | ahhh duh... |
21:49.06 | jkroon | [TK]D-Fender, if you really want to spend time I can quickly try and get two traces for you, but I'm actually about off to bed ... was just wondering whether anybody would be brave enough to venture a guess or two. |
21:49.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
21:49.54 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.77.124) |
21:54.31 | [TK]D-Fender | jkroon: Asking us to guess with no info is kinda retarded |
21:55.35 | jkroon | probably. busy trying to get two traces. but keep on getting BUSY between china and za ... so may take a few minutes. |
21:58.44 | drfreeze | Anyone know if the ringback feature of parking works? |
21:58.50 | drfreeze | I'm getting the following error: Spawn extension (inbound, 4752, 2) exited non-zero on 'Parked/DAHDI/2-1<ZOMBIE>' |
21:59.20 | drfreeze | this is immediately following the parking of the call |
22:03.17 | hardwire | meh. |
22:03.29 | hardwire | pokes manxpower in the eye. |
22:10.35 | bmoraca | hardwire: did you ever get your packet routing problem fixed? |
22:10.53 | hardwire | in a way. |
22:11.07 | hardwire | I think it's to complex for common protocols. |
22:11.12 | hardwire | and methods |
22:11.13 | hardwire | and such. |
22:11.23 | hardwire | but.. that's me in a nutshell. |
22:11.32 | hardwire | or maybe out.. |
22:11.35 | hardwire | I think out of the nutshell? |
22:11.37 | hardwire | is that right? |
22:11.39 | hardwire | heh. |
22:12.03 | bmoraca | lol |
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22:27.51 | drfreeze | Anyone know the magic handshake to get call parking call returns working? |
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22:31.56 | hardwire | drfreeze: tried konami code? |
22:32.34 | drfreeze | hardwire: what's that? |
22:32.59 | Chainsaw | Well someone just gave away their age. |
22:33.05 | hardwire | â â â â â â â â B A |
22:33.47 | hardwire | Chainsaw: I believe you, I, and drfreeze just did. |
22:34.11 | jkroon | [TK]D-Fender, http://pastebin.co.za/44186 - in some cases I see CLI in others I don't. I don't see anything there that can possibly make a difference. |
22:34.22 | Chainsaw | hardwire: Anyhow, blow the dust out and plug it back in. That always works. |
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22:34.48 | Chainsaw | hardwire: http://imgs.xkcd.com/comics/nintendo_surgeon.png |
22:35.01 | jkroon | that particular trace did show the CLID ... retrying now to see if I can get one without. |
22:35.02 | mascool | hello everyone, quick question about dahdi |
22:35.21 | mascool | dahdi show status returns this: DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFIGUR 0 0 0 |
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22:35.24 | mascool | is this ok ? |
22:35.41 | mascool | i don't have any PRI/FXO etc interface cards |
22:35.58 | jkroon | that's fine. |
22:36.22 | jkroon | just make sure you pass -I to asterisk startup or have in asterisk.conf inside [options] internal_timing=yes |
22:36.47 | mascool | oh sweet I didn't know about that |
22:36.48 | mascool | thanks |
22:37.07 | hardwire | Chainsaw: heh |
22:39.51 | jkroon | mascool, much better sip quality :p |
22:40.03 | jkroon | i'm of the opinion they should make it the default ... but alas. |
22:40.07 | mascool | jkroon: in what sense? |
22:40.20 | mascool | i've been running a box without that setting for 1 year and i've had no problems |
22:40.38 | jkroon | well, your timers are more accurate, instead of locking transmits onto receives it actually generates it's own timing. |
22:41.01 | jkroon | mascool, check again. chances are you've been passing -I without knowing it. |
22:41.29 | jkroon | [TK]D-Fender, if by any chance you're still willing to make a guess or two, a bit more detailed: http://pastebin.co.za/44187 |
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22:41.49 | mascool | nope, asterisk in init.d doesn't have that option turned on |
22:44.30 | mascool | jkroon: any other place I should check ? |
22:44.58 | jkroon | mascool, i find it most reliable to run "ps axf | grep asterisk" or to even check in /proc/$(pidof asterisk) |
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22:45.54 | mascool | hmm ok and how can I determine if I passed -I from there ? |
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22:48.32 | hardwire | do sip reinvites hit CEL? |
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22:56.57 | hardwire | 222 users in here. |
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23:07.49 | drfreeze | Help |
23:07.57 | drfreeze | Is there a bug in call parking? |
23:08.12 | drfreeze | My parked calls never ring back |
23:08.13 | drfreeze | Parked DAHDI/1-1 on 701@parkedcalls. Will timeout back to extension [inbound] 4753, 4 in 5 seconds |
23:09.22 | grandpapadot | drfreeze: pastebin your features.conf and extensions.conf and I'll take a look |
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23:11.09 | generalhan | speaking of bugs, i still havent had a response to the "bug" that i submitted. i kinda expected to get something like "you are a retard, piss off" within minutes of submitting it ! lol |
23:15.56 | [TK]D-Fender | [19:07]<drfreeze>Is there a bug in call parking? <- what kind of aimless bait is that? |
23:16.00 | drfreeze | http://gist.github.com/174298 |
23:16.14 | generalhan | lol |
23:16.23 | drfreeze | [TK]D-Fender: all I see from google is problems and bugs, but no fixes |
23:17.04 | [TK]D-Fender | drfreeze: You ask a wide-scope question with no sense of VERSION or debug to share.... |
23:17.09 | drfreeze | grandpapadot left before I could paste |
23:17.14 | [TK]D-Fender | drfreeze: That is flamebait, pure and simple. |
23:17.25 | drfreeze | asterisk-1.4.24.1 |
23:17.41 | [TK]D-Fender | drfreeze: Automatically the first answer is UPGRADE |
23:17.45 | drfreeze | with DAHDI, and TE121, if that matters |
23:17.47 | [TK]D-Fender | drfreeze: You're clearly behind. |
23:17.57 | drfreeze | can't, there is a bug in the latest |
23:18.00 | [TK]D-Fender | drfreeze: Still doesn't show us configs, or calls |
23:18.06 | drfreeze | a documented bug |
23:18.17 | [TK]D-Fender | drfreeze: And I promise you there are bugs in that version too. |
23:18.20 | drfreeze | [TK]D-Fender: see gist paste above for conf files |
23:18.48 | [TK]D-Fender | drfreeze: where's the call? |
23:19.29 | drfreeze | incoming calls are answered and parked via #700 |
23:19.42 | drfreeze | they are successfully parked, but never time out and ring back |
23:19.43 | [TK]D-Fender | drfreeze: where's the call? <---------------- |
23:19.56 | drfreeze | what do you mean? |
23:20.12 | [TK]D-Fender | drfreeze: Don't understand what a CALL is>? |
23:20.22 | [TK]D-Fender | drShow me you parking a call and the parking screw up <- |
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23:21.12 | drfreeze | [TK]D-Fender: updated the gist |
23:21.15 | drfreeze | please refresh |
23:21.36 | drfreeze | also, moh is stopped when parking |
23:21.56 | [TK]D-Fender | drfreeze: drfreeze the ENTIRE call. |
23:22.11 | drfreeze | one min |
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23:23.41 | drfreeze | [TK]D-Fender: updated gist with all call info with debug and verbose set to 30 |
23:25.12 | [TK]D-Fender | drfreeze: Where are you calling the parking app? I don't see that in there. |
23:25.53 | [TK]D-Fender | drfreeze: Also what are you doing to park the call? What phone are you doing it on? |
23:25.55 | drfreeze | I don't call it directly |
23:26.13 | drfreeze | #700 puts it in the parking lot. I can retrieve it by dialing 701 |
23:26.21 | drfreeze | Polycom 550 |
23:26.57 | [TK]D-Fender | drfreeze: don't use "#" DTMF features are ridiculous. use the native transfer. |
23:27.49 | drfreeze | ok |
23:29.13 | drfreeze | [TK]D-Fender: ok. I'll implement that and give it a try. |
23:30.29 | ZX81 | ~seen oej |
23:30.33 | infobot | oej <n=olle@ns.webway.se> was last seen on IRC in channel #asterisk-dev, 4h 8m 47s ago, saying: '(Can't access the servers now)'. |
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23:35.08 | freenose | What's a good headset to use with polycom 331? |
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23:41.48 | drfreeze | [TK]D-Fender: using the api seems to require a blind xfr, which, I think is '#1'. Do you suggest another method? |
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23:44.50 | [TK]D-Fender | drfreeze: API? What are you talking about.. you were using #700 from a PHONE... |
23:45.02 | [TK]D-Fender | drfreeze: use the bloody TRANSFER button on the phone. |
23:45.09 | [TK]D-Fender | is off for a while. |
23:50.46 | watchy | tk: anyway to get a polycoms model from dhcp? |
23:55.18 | Qwell | watchy: you could sorta guess by MAC, I bet |
23:55.54 | drfreeze | I'm looking at the examples on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce |
23:56.09 | drfreeze | and it seems this variable has changed names: BLINDTRANSFER |
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