| 00:00.04 | bcrisp | maybe i can set up the option prior to joining the queue |
| 00:00.21 | Kobaz | so you can dial a Local channel, which answers, and goes to the queue that plays music |
| 00:00.29 | Kobaz | and then, through the dial, now you have your feature codes |
| 00:00.46 | bcrisp | interesting |
| 00:01.24 | Kobaz | and then there's the DYNAMIC_FEATURES channel variable |
| 00:01.43 | Kobaz | you need to set it, to cointain a list of feature codes that are available to the caller before dial()ing |
| 00:01.59 | bcrisp | wow |
| 00:02.02 | bcrisp | nice |
| 00:02.18 | Kobaz | i forgot how i found that, but it took me a really long time to get feature codes working right |
| 00:02.26 | Kobaz | but you should have all the pieces you need |
| 00:02.32 | flux_control | If I'm receiving a call and want to connect it to a SIP extension, I've seen most dialplans using Dial for that. |
| 00:02.34 | Kobaz | features, local channel, sit in queue |
| 00:02.58 | flux_control | Would there be any advantage/disadvantage to using Originate or Bridge instead? |
| 00:03.01 | Kobaz | it's a SIP *device* |
| 00:03.04 | bcrisp | studying the dial flags |
| 00:03.19 | Kobaz | extensions are dialplan jump points |
| 00:03.29 | flux_control | Kobaz: Right, tomatoe, tomato. ;) |
| 00:03.37 | flux_control | heh |
| 00:03.45 | Kobaz | not really |
| 00:04.00 | Kobaz | if you talk to tk-fender he will give you a hard time about that |
| 00:04.00 | flux_control | Anyway, for connecting to the device, would there be any benefits or not for using Originate or Bridge? |
| 00:04.01 | Kobaz | heh |
| 00:04.14 | Kobaz | originate/bridge are not the tools for that |
| 00:04.29 | Kobaz | originate makes a *new* call, with *two new* endpoints |
| 00:04.40 | Kobaz | bridge will link a caller to an existing channel |
| 00:05.01 | flux_control | Kobaz: Well, I could set up a local device, connect the incoming call to local, and then bridge the local device to the SIP device, right? |
| 00:05.14 | Kobaz | you could |
| 00:05.36 | Kobaz | i actually do that for my custom queue code that I wrote, but it's for an entirely different purpose |
| 00:05.36 | flux_control | Just wondering if that would enable any tricky features or not. |
| 00:05.49 | Kobaz | it's just extra work to do what Dial() does anyway |
| 00:05.54 | flux_control | I see. |
| 00:06.02 | Kobaz | here's what you would need, to use originate |
| 00:06.51 | Kobaz | answer the call, do an originate() to the destination, and a source of a local dialplan context that runs bridge() with the channel that just came in |
| 00:07.07 | flux_control | Right |
| 00:07.30 | Kobaz | if all you want to do |
| 00:07.33 | Kobaz | is connect a to b |
| 00:07.47 | Kobaz | it's a lot of extra work |
| 00:07.59 | flux_control | Yeah |
| 00:08.37 | flux_control | Would it have any benefits for nat traversal? |
| 00:08.43 | Kobaz | not at all |
| 00:08.43 | flux_control | I doubt it, but just wondering. |
| 00:08.47 | flux_control | OK |
| 00:08.53 | flux_control | Thanks :) |
| 00:09.00 | Kobaz | it uses the same built in nat stuff that is part of asterisk |
| 00:09.25 | Kobaz | any IP device to any other IP device, will use whatever asterisk has built in for nat |
| 00:09.31 | Kobaz | no matter what dialplan shenanagins you use |
| 00:10.03 | flux_control | Gotcha |
| 00:11.20 | Kobaz | time to make the donuts |
| 00:11.42 | flux_control | 6am? ;p |
| 00:11.46 | Kobaz | heh |
| 00:11.48 | Kobaz | 7:11pm |
| 00:12.05 | flux_control | Ah, EDT. |
| 00:12.07 | flux_control | :) |
| 00:12.17 | Kobaz | need to write two more features by tomorrow |
| 00:13.50 | flux_control | I wonder if Transfer()ing incoming PSTN-over-SIP calls will work better for me than Dial()ing them... |
| 00:14.04 | flux_control | For the ipkall and sipbroker, that is.. |
| 00:14.47 | Kobaz | pstn over what type of device |
| 00:14.51 | Kobaz | dahdi? sip? |
| 00:15.19 | Kobaz | dahdi -> transfer() -> sip doesn't gain you anything... it's the same as a dial() |
| 00:15.50 | flux_control | No dahdi. |
| 00:15.56 | Kobaz | sip -> transfer() -> sip ... (if reinvite=yes in your config), it will link the endpoints directly together, and take asterisk out of the picture |
| 00:16.39 | Kobaz | but... if you want to record calls, then you can't do a remote transfer like that |
| 00:16.47 | Kobaz | since asterisk needs to be in the loop |
| 00:16.53 | flux_control | I try receiving calls from PSTN via sipbroker or ipkall, and the local (meaning either the softphone or asterisk, I can't tell which) terminates the call after about 20 secs. |
| 00:17.11 | flux_control | However, the PSTN device stays connected until it receives a timeout from ipkall or sipbroker. |
| 00:17.30 | Kobaz | transfer() won't help you |
| 00:17.43 | flux_control | I'm just trying to figure out how the local can hangup but that doesn't reach the remote. |
| 00:17.47 | Kobaz | best thing is to wait for tomorrow, find [TK]D-Fender, and post up your sip debug |
| 00:18.14 | flux_control | I'll try that, but probably won't be able to get on tomorrow. |
| 00:18.17 | flux_control | Busy day. |
| 00:18.29 | *** join/#asterisk mr_ian_ (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 00:18.30 | flux_control | Thanks though :) |
| 00:19.02 | Kobaz | he's really good with finding sip issues... i'm getting there |
| 00:19.07 | Kobaz | np |
| 00:19.09 | Pan3D | post it here anyway, maybe someone coming back to their screen will take a peak |
| 00:19.17 | Kobaz | yeah, that too |
| 00:19.24 | Kobaz | asterisk hangs on to the call? |
| 00:19.28 | Kobaz | the soft phone hangs up? |
| 00:19.32 | Kobaz | is that what happens... |
| 00:19.51 | flux_control | In the softphone, I get a call terminated, and in * it shows that the call gets terminated. |
| 00:20.06 | flux_control | The PSTN I call with (my cell) shows that it's still connected |
| 00:20.09 | Kobaz | your sip provider has crappy disconnect supervision |
| 00:20.15 | flux_control | Probably |
| 00:20.22 | flux_control | This was with ipkall and sipbroker. |
| 00:20.29 | Kobaz | heh. not probably... that's what's going on |
| 00:20.41 | Kobaz | i thought your soft phone was timing out or something |
| 00:20.51 | flux_control | Not sure why I got the disconnects in the first place, I'm guessing probably also the provider sucks. |
| 00:20.54 | Kobaz | if your asterisk channel is gone... ipkall gets the hangup too |
| 00:21.17 | Kobaz | if ipkall handles the hangup wrong, your other phone will be left on the call |
| 00:21.27 | flux_control | The bastards! |
| 00:21.28 | flux_control | heh |
| 00:21.37 | Kobaz | so you have two problems? |
| 00:21.41 | Kobaz | itsp and soft phone? |
| 00:21.44 | *** join/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) |
| 00:21.50 | flux_control | Not sure |
| 00:22.00 | kam187 | well i fixed my problem with chan_sebi :) |
| 00:22.09 | flux_control | I saw retransmits in the SIP debug, but I'm not sure if that's normal. |
| 00:22.19 | kam187 | the option (modem driver) uses an out buffer of only 64 bytes |
| 00:22.24 | kam187 | had to rexompile the kernel module |
| 00:22.26 | flux_control | And if not normal, I'm not sure what the cause is. |
| 00:22.30 | kam187 | (if anyone cares) :) |
| 00:22.43 | Kobaz | something crappy in the networking or the endpoints |
| 00:22.55 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 00:22.59 | Kobaz | kam187: iaxmodem? |
| 00:25.25 | flux_control | Kobaz: If I use a if (ChanIsAvail(SIP/exten)) { Answer(); Dial(SIP/exten); Hangup(); }, is that sufficient, or do I need to specifically check ${AVAILSTATUS} or something? |
| 00:26.07 | Kobaz | applications don't have return codes |
| 00:26.18 | Kobaz | you need to check the AVAILSTATUS variable |
| 00:27.55 | flux_control | Kobaz: Cool, thanks a bunch! |
| 00:28.07 | flux_control | I'll let you get on with making zee donuts :) |
| 00:28.18 | Kobaz | hehe |
| 00:37.37 | *** join/#asterisk CaT[t3] (n=cat@nessie.weebeastie.net) |
| 00:38.19 | flux_control | Man, I wish the values for ChanIsAvail variables were in the docs... |
| 00:38.43 | flux_control | Aha, they're on the wiki. |
| 00:38.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:43.33 | CaT[t3] | has anyone here played with the srtp asterisk branch? |
| 00:46.58 | *** join/#asterisk dongs (i=1000@l212168.ppp.asahi-net.or.jp) |
| 00:47.16 | dongs | can someone please explain to me why the fuck my SIP calls get disconencted after 420 seconds. this is not fucking funny. |
| 00:47.51 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 00:48.29 | chuckf | dongs: do you have a firewall between the phone and the asterisk box? |
| 00:48.42 | dongs | .. no. |
| 00:48.57 | dongs | its asterisk that drops calls, not the phone. |
| 00:49.38 | chuckf | if you dial a local extention does it drop as well after that time? |
| 00:49.43 | flux_control | Is it asterisk, or your provider? |
| 00:50.00 | flux_control | Have you tried doing a call between internal phones, only using asterisk? |
| 00:50.14 | flux_control | I guess that's the same as what chuckf just asked. :) |
| 00:52.21 | dongs | the only thing that changed was upgrading asterisk from <something old> to <something newer> |
| 00:53.14 | chuckf | but that doesn't answer what I asked |
| 00:53.24 | *** join/#asterisk ming_zym (n=ming_zym@114.251.86.0) |
| 00:54.12 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:54.59 | dongs | i have no way of testing what you've said, as all internal phones are connected to a real key system, not this voip junk. |
| 00:55.08 | dongs | and yes, they can talk extension to extension all they want. |
| 00:55.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:59.03 | dongs | is there any sensible way to log sip debug output to uh,, a file? |
| 00:59.26 | dongs | or is that too hard in 2009 |
| 00:59.29 | flux_control | dongs: Can you try having two sip soft phones on the same lan call each other and see if they drop at the same time? |
| 01:00.14 | flux_control | Hmm... why is ChanIsAvail(); if ("${AVAILSTATUS" |
| 01:00.18 | dongs | dont have any softphones. all asterisk is used for is to turn 2 voip lines into ISDN so they can go into a real key system. |
| 01:00.22 | flux_control | oops, darn enter key |
| 01:00.57 | flux_control | Why is ChanIsAvail(); if ("${AVAILSTATUS" = "AST_DEVICE_NOT_INUSE") not working for a device which is definitely not in use, but registered? |
| 01:01.36 | flux_control | dongs: Do you have phones on those 2 voip lines? |
| 01:01.48 | dongs | what do you mean phones |
| 01:01.53 | flux_control | Any kind of phone |
| 01:02.08 | flux_control | Or would it be possible to hook phones up to those two lines? |
| 01:02.17 | flux_control | If so, then you can test them. |
| 01:02.34 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 01:03.05 | flux_control | Then, if you establish those calls work, there's likely a problem going between the voip and the key system, but at least the rest of asterisk would be working. |
| 01:03.19 | flux_control | In other words, you would have narrowed down the problem. |
| 01:03.58 | flux_control | If you can't get any phones working where asterisk is the only intermediary, and it isn't transferring across systems, then your asterisk is likely to be quite broken (or at least the setup). |
| 01:04.50 | CaT[t3] | is having issues with it (srtp) and can't figure it out. currently just using rtpw from libsrtp to test it and is getting no output on the receiving side, encryption or no. should he be? |
| 01:10.52 | dongs | sorry, 260 seconds. 4 minutes 20 seconds. there is nowhere anything about '260' seconds in any config file. |
| 01:12.24 | flux_control | dongs: That says nothing about what chuckf and I are suggesting. |
| 01:14.26 | dongs | .. how do i tell from ChangeLog which version a change applies to? |
| 01:17.29 | dongs | .. anyone? |
| 01:19.43 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:19.43 | dongs | im going to upgrade from 1.6.1.5 to 1.6.1.11 and hope it solves the problem, as unlikely as that is. |
| 01:20.01 | dongs | can I copy some file from my old 1.5 build so I dont have to go through over 9k pages of menuconfig again? |
| 01:20.05 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:21.19 | ChannelZ | Kumbang and dongs. Too easy. |
| 01:21.27 | ChannelZ | :) |
| 01:22.09 | dongs | real funny. do you have any actual useful input to my questions? |
| 01:22.35 | dongs | 1) how to find out what asterisk version a entry in ChangeLog refers to |
| 01:22.45 | dongs | 2) How to copy build settings from older version to newer |
| 01:23.35 | ChannelZ | Look for "Release Asterisk x.x.x.x" and do the math |
| 01:24.24 | dongs | that would work, if they actually followed that through. |
| 01:24.28 | ChannelZ | use your menuselect.makeopts and menuselect.makedeps probably (read much?) |
| 01:24.33 | *** join/#asterisk [netman] (n=netman@223.Red-88-19-164.staticIP.rima-tde.net) |
| 01:24.34 | dongs | it jumps from * Release Asterisk 1.6.1.0-rc3 |
| 01:24.36 | dongs | to 1.6.1.7 |
| 01:27.36 | Katty | yawn. |
| 01:28.43 | ChannelZ | gives Katty over-the-counter wake-up drugs |
| 01:31.57 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 01:40.48 | ChannelZ | "LibPRI is a library that encapsulates the protocols used to communicate over ISDN Primary Rate Interfaces (T1, E1, J1). LibPRI is a dependency for Asterisk and Zaptel if PRI signaling is used. For a more detailed explanation, check out the About LibPRI page." |
| 01:40.57 | ChannelZ | clicks |
| 01:41.05 | ChannelZ | "LibPRI is an open source library that implements signaling for ISDN-PRI and ISDN-BRI." |
| 01:41.14 | ChannelZ | I'm pretty sure that's less information. |
| 01:41.56 | *** join/#asterisk arisi (n=user@129.21.55.235) |
| 01:43.17 | arisi | I have a fresh asterisk install on a debian lenny machine, and I have followed the instructions at http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client to do basic setup, but whenever I try to call myself, it immediately returns call completed. Does anyone know why? |
| 01:43.25 | dongs | i wonder if I should jsut switch to SER or something. |
| 01:43.31 | dongs | stuff that just works. |
| 01:43.44 | arisi | I am just trying to do a simple sip server |
| 01:45.55 | *** join/#asterisk arisi (n=user@129.21.55.235) |
| 01:46.33 | dongs | crap. i think i had to patch some stupidity in 1.6.1.5 to make it work but i forgot what it was. |
| 01:46.52 | ChannelZ | whose stupid now? |
| 01:47.13 | ChannelZ | arisi: not without seeing some configs |
| 01:48.08 | dongs | oh right, including taking out -g3 from Makefile |
| 01:48.28 | dongs | i wonder whose great idea it was |
| 01:48.34 | dongs | to deliver release product with -g3 |
| 01:49.00 | dongs | http://bcas.tv/paste/results/eP6Ykf83.html and then idiocy like this |
| 01:49.08 | dongs | 'lets "FIX" one device and break the rest' |
| 01:51.09 | arisi | ChannelZ, I might have gotten it working, actually |
| 01:51.14 | arisi | I was expecting more of a fight |
| 01:51.35 | Katty | hi |
| 01:51.41 | Katty | i've got spaghettis |
| 01:52.01 | ChannelZ | mmmm |
| 01:52.27 | Katty | yesh |
| 01:52.28 | *** join/#asterisk lordmortis (n=lordmort@203-59-175-226.dyn.iinet.net.au) |
| 01:52.46 | *** join/#asterisk mintos (n=mvaliyav@117.206.17.112) |
| 01:54.12 | dongs | i see the uniden hack is still in 1.11 unchanged. |
| 01:54.40 | ChannelZ | dongs: maybe you should go stomp your feet in #asterisk-bugs |
| 01:55.27 | *** join/#asterisk lordmortis (n=lordmort@gateway.qsix.com.au) |
| 01:55.27 | Katty | stomps feet |
| 01:55.41 | dongs | yes because a total of 0 more people will see it there than here. |
| 01:55.54 | ChannelZ | (that's the point) |
| 01:59.33 | flux_control | arisi: You're calling *yourself*? |
| 01:59.43 | flux_control | As in, from your ekiga client, calling your ekiga client? |
| 02:06.09 | dlynes | dongs: have you tried pcapsipdump for the debug to one log file? |
| 02:06.11 | *** join/#asterisk ticoit (n=ticoit@201.191.151.147) |
| 02:06.34 | dongs | uh what. why cant I just tell asterisk to log to a file? |
| 02:06.42 | dlynes | dongs: pcapsipdump allows you to dump every single sip conversation (both legs of it) to a separate file |
| 02:07.02 | dlynes | dongs: and it's in pcap/wireshark format |
| 02:07.19 | Katty | restreaming critter cam. |
| 02:07.25 | Katty | infobot: critter cam |
| 02:07.26 | infobot | methinks critter cam is http://www.ustream.tv/channel/squirrel-critter-cam |
| 02:07.33 | dongs | why would i want to re-dump something asterisk already prints. |
| 02:07.40 | *** join/#asterisk CrazyTux (n=brandon@70-138-119-70.lightspeed.hstntx.sbcglobal.net) |
| 02:08.11 | dlynes | dongs: debug => debug (logger.conf) |
| 02:08.11 | CrazyTux | Hey guys I'm using AMI and I am sending an "Originate" action and I want to get the returned result of the call i.e. answered, failed, etc |
| 02:08.37 | CrazyTux | How exactly is this done, I am looking the voip-info wiki on this however it says it is possible it does not give the slightest clue as to how it is determined |
| 02:08.53 | dlynes | CrazyTux: agi? |
| 02:08.57 | CrazyTux | AMI |
| 02:08.59 | CrazyTux | not AGI |
| 02:09.02 | CrazyTux | Asterisk Manager Interface |
| 02:09.04 | dlynes | CrazyTux: agi |
| 02:09.13 | CrazyTux | dlynes: your saying it is not possible with AMI? |
| 02:09.22 | dlynes | CrazyTux: i don't think you can do it with ami, personally |
| 02:09.31 | CrazyTux | Documentation is saying you can |
| 02:09.41 | dlynes | CrazyTux: maybe you can, then |
| 02:10.01 | CrazyTux | "Using Async leads to an OriginateResponse event which contains the failure reason if any. Reason may be one of the following: " |
| 02:10.15 | CrazyTux | 0 = no extension, 1 = no answer, 4 = answered, 8 = congested |
| 02:11.42 | *** join/#asterisk jmacz (n=jmacz@190.25.7.149) |
| 02:12.35 | *** join/#asterisk mokmeister (n=mokmeist@109.78.70.51) |
| 02:14.34 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 02:16.01 | *** part/#asterisk flux_control (n=flux@sourcemage/wizard/flux) |
| 02:17.58 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 02:18.03 | dlynes | CrazyTux: where did you find that info? |
| 02:18.12 | CrazyTux | dlynes: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 02:19.40 | dlynes | CrazyTux: it's not working for you? |
| 02:19.52 | dlynes | CrazyTux: a lot of stuff on that website is outdated, or obsolete |
| 02:21.15 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 02:22.41 | ChannelZ | wanders off to make a pizza |
| 02:29.57 | Katty | follows ChannelZ |
| 02:30.17 | *** join/#asterisk CyberPony (n=CyberPon@cpe-071-075-195-085.carolina.res.rr.com) |
| 02:32.30 | *** join/#asterisk mintos (n=mvaliyav@117.206.17.98) |
| 02:37.53 | jblack | I'm making pizza too. |
| 02:38.55 | dongs | whats the junk in sip debug that looks like i: blah c: blah t: blah v: blah k: blah etc |
| 02:39.06 | dongs | is that actual data over the wire or they're formatting it |
| 02:41.43 | DocAwesome | dongs: it's short form headers |
| 02:41.56 | DocAwesome | at least that's what it looks like -- hard to tell without seeing the actual header |
| 02:42.16 | dongs | well, where are tehy getting parsed. |
| 02:42.45 | DocAwesome | question does not parse |
| 02:43.07 | lost_soul | I'm wondering anyone have experience with pjsua? I'm trying to figure out if it's possible to get some type of media file to play when a call is coming in so their is some way of knowing |
| 02:43.46 | lost_soul | I know it's off topic, and for that I'm sorry. But this would be the most appropriate channel I could think of to ask in |
| 02:43.51 | dongs | http://bcas.tv/paste/results/dDgAxj49.html |
| 02:44.16 | dongs | i want to check if asterisk is failing to parse this stuff due to spaces, or other idiotic formatting etc. so where are those getting parsed. |
| 02:44.24 | *** part/#asterisk CyberPony (n=CyberPon@cpe-071-075-195-085.carolina.res.rr.com) |
| 02:44.32 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 02:44.44 | dongs | specifically the k: part. |
| 02:44.48 | dongs | and x: |
| 02:45.12 | CrazyTux | dlynes: is there a new resource for asterisk documentation/' |
| 02:45.24 | CrazyTux | dlynes: it is not working because it does not tell me how to retrieve OriginateResponse |
| 02:45.43 | CrazyTux | it says its an event, but I am not seeing it |
| 02:45.44 | DocAwesome | dongs: not sure -- look at the sip spec. That's short form headers, which should work, but I don't know too many end points that use it |
| 02:46.48 | leifmadsen | dongs: they are getting parsed in chan_sip.c |
| 02:47.21 | dongs | leifmadsen: no kidding, thank you sir, proceeding to read 900kbytes of chan-sip line by line |
| 02:47.38 | leifmadsen | not sure what other answer you were expecting |
| 02:47.43 | dongs | ... |
| 02:48.26 | dongs | i expected you to go into a multi-line diatribe on advantages of installing ubuntu. |
| 02:49.33 | dongs | are you saying tehre's a fairly high chance that those arent even getting parsed? |
| 02:49.59 | dongs | "since nobody uses them, fuck'em!" |
| 02:51.27 | *** join/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 02:51.45 | minotaur01 | hey everyone |
| 02:52.10 | leifmadsen | dongs: I see no reason why any distro would be better suited than another when running linux |
| 02:52.21 | leifmadsen | that would have absolutely no effect on your parsing issue |
| 02:52.28 | dongs | precisely. |
| 02:52.34 | leifmadsen | the parsing is of a SIP header, which is done in chan_sip.c |
| 02:52.43 | leifmadsen | you're a difficult person to attempt to help, so I'll just go read instead |
| 02:53.10 | dongs | so your confusion regarding my question of WHERE those headers are parsed seems strange. |
| 02:54.30 | leifmadsen | you want me to point out the exact line in chan_sip? sorry |
| 02:55.58 | jblack | You have to be pretty bad to get on Leif's nerves. |
| 02:56.36 | jaytee | wonders if some people take lessons in being a dick or whether it just comes naturally |
| 02:56.53 | dongs | the part parsing x: would be parse_session_expires(). I see nowehre calling it that isnt related to 'get_header("Session-Expires")' |
| 02:57.07 | dongs | does chan_sip do some substitute of short->long headers during inital packet parse? |
| 02:57.43 | jblack | dongs: I'll be more clear... The question that you are asking is of such complexity, that not even the guy who wrote the book on asterisk (literally), knows the answer. As such, he's referring you to the source code, which is the only place for which that answer really exists. |
| 02:58.02 | dongs | thats where im looking at, genius. |
| 02:58.17 | dongs | im asking some basic questions of how it works so I can understand if im looking in the right place or not. |
| 02:59.46 | jblack | Maybe you could hit up digium for a support call. |
| 02:59.55 | jblack | There's a few guys there that know the sip codebase pretty well. |
| 02:59.57 | jaytee | or ask in #asterisk-dev |
| 03:03.28 | *** join/#asterisk chendy (n=chatzill@119.122.244.182) |
| 03:03.29 | jblack | the problem with #asterisk-dev is that a lot of the people that camp there camp here too, and they'll see he acted like a dick here. |
| 03:03.38 | dongs | indeed. |
| 03:03.52 | dongs | so asking there is just as pointless as asking here. |
| 03:04.31 | lost_soul | which is why you should be thankful for the help people try to give :) |
| 03:04.54 | dongs | the moment any actual help is given i'll be all over it. |
| 03:05.17 | jaytee | and I used to think my ex whined alot |
| 03:05.20 | jblack | I'll reiterate my suggestion; purchase support at digium. |
| 03:05.26 | lost_soul | with that attitude, I'm sure nobody would mind if you hold your breath while waiting |
| 03:05.26 | dlynes | lost_soul: he's been acting like a dick since he got in here...I have no idea what his question even is....last five screens are nothing but bitching and complaining |
| 03:05.28 | toresbe | The moment you start acting like a less friendly version of the Comic Book Guy I think you might risk getting it. |
| 03:05.51 | lost_soul | dlynes: I noticed, I've been lurking in here for awhile |
| 03:06.29 | dlynes | lost_soul: but maybe he doesn't have a question, and he just wants to bounce his frustration off the wall |
| 03:06.51 | jblack | two, maybe three people will know that answer. You sould check the svn logs |
| 03:06.51 | dlynes | or maybe he's a callweaver developer...heaven forbid... |
| 03:07.01 | jblack | trixbox. :) |
| 03:07.30 | file | get_header takes into account compact headers. |
| 03:07.42 | dongs | and? |
| 03:07.47 | file | and what? |
| 03:07.48 | lost_soul | dlynes: having issues is nobody's fault, it may not even be something he's doing wrong. So taking it out on people that help people day in and day out is hardly a productive use of time |
| 03:07.58 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 03:07.59 | file | there's an internal structure of mappings between compact headers and the real header |
| 03:08.06 | dongs | so when im callingget_Header("Session-Expires") it will return either that one or x:? |
| 03:08.10 | file | when a header is requested it is converted from full header to the compact header |
| 03:08.15 | file | (if need be) |
| 03:08.37 | file | Session-Expires is equivalent to x |
| 03:08.38 | *** join/#asterisk MaliutaLap (n=biteme@96.53.148.53) |
| 03:08.39 | dongs | ya, i see the map table now. |
| 03:08.40 | dongs | right. |
| 03:09.14 | jaytee | omg! this is funny! http://blogs.riverfronttimes.com/gutcheck/2009/12/audio_woman_calls_911_to_get_burger_king_st_louis_food_blog_120709.php |
| 03:09.29 | dongs | at which setting hmm. |
| 03:09.45 | dongs | file: when does stuff from ast_debug(2, etc get printed to console/log? |
| 03:09.47 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 03:10.04 | dongs | core set debug atleast bla? |
| 03:10.31 | file | yes, and if debug is set to go to the log or console |
| 03:10.43 | dongs | wheres that part |
| 03:10.53 | dlynes | dongs: logger.conf (console => debug) |
| 03:11.12 | file | I'll answer questions that the vast majority of the rest of the people in here can't answer but I draw the line at basic stuff. |
| 03:11.36 | dongs | i was looking at logger.conf already. |
| 03:11.39 | dongs | thats setup. |
| 03:11.48 | dongs | i dont see them being printed though. |
| 03:12.00 | lost_soul | jaytee: lol, people in my area I could see doing that |
| 03:12.15 | jaytee | I loved the audio track |
| 03:12.35 | dlynes | dongs: are you doing core set debug 100 and sip set debug on? |
| 03:12.41 | lost_soul | I didn't listen to that part yet, just read the text |
| 03:12.50 | jaytee | Caller: You're supposed to protect me... Dispatcher: What are we protecting you from, a wrong cheeseburger? |
| 03:12.58 | dongs | dlynes: yea. |
| 03:14.55 | dlynes | dongs: if you've got that, and you've made sure console isn't commented out, and it has debug as an option, and you did a logger reload, you should be getting it |
| 03:15.06 | dlynes | dongs: otherwise you've got bigger issues, and it ain't asterisk |
| 03:15.18 | dongs | i uncommented full, and im watching that file |
| 03:15.28 | dongs | and restarted , instead of loger reload so that should be good. |
| 03:15.37 | dlynes | dongs: that should work, too |
| 03:15.37 | dongs | anyway, still reading through the junk to see if waht i want got printed |
| 03:15.56 | dlynes | dongs: but now you're talking about the full log, not console |
| 03:16.10 | dlynes | dongs: are you wanting it in a file, or dumped to the screen? |
| 03:16.38 | *** join/#asterisk jmworx___ (n=jeval@mail.octasic.com) |
| 03:16.38 | dongs | file. i think it worked, there was actually more junk in file than on screen. im getting DEBUG[] stuff. |
| 03:17.29 | dlynes | file's filled with junk? He's so nice to you, and then you insult him...sheesh |
| 03:17.49 | dongs | waht i narrowed my 4min20second disconnect to is session starts with sessiontimer active, then it becomes inactive after 2min20sec |
| 03:17.53 | dongs | followed by disconenct 2 min later. |
| 03:20.24 | dlynes | IBM's newest mainframe is all Linux: http://hardware.slashdot.org/story/09/12/09/2329211/IBMs-Newest-Mainframe-Is-All-Linux?from=rss&utm_source=feedburner&utm_medium=feed&utm_campaign=Feed%3A+Slashdot%2Fslashdot+%28Slashdot%29 |
| 03:20.24 | dlynes | cool |
| 03:20.24 | minotaur01 | hey dlynes |
| 03:20.30 | dlynes | You can get into an ultra super cool asterisk box for only $212K |
| 03:20.30 | dlynes | cheaper than a house |
| 03:20.35 | dlynes | hey mino |
| 03:20.36 | file | I'm sure I could expense that. |
| 03:20.37 | *** join/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) [NETSPLIT VICTIM] |
| 03:20.37 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) [NETSPLIT VICTIM] |
| 03:20.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 03:20.38 | *** join/#asterisk path (i=path@server1.bshellz.net) [NETSPLIT VICTIM] |
| 03:20.38 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) [NETSPLIT VICTIM] |
| 03:20.38 | *** join/#asterisk cmendes0101 (n=none@96.247.7.25) [NETSPLIT VICTIM] |
| 03:20.45 | minotaur01 | lol thats cheap |
| 03:20.45 | dlynes | well, unless you live in detroit, of course |
| 03:21.02 | *** join/#asterisk _bugz_ (n=bugz@adsl-76-194-244-221.dsl.lsan03.sbcglobal.net) [NETSPLIT VICTIM] |
| 03:21.16 | minotaur01 | i wonder how many phone calls a 212K box can handle |
| 03:21.40 | dlynes | It's a dual processor, and both processors are specifically designed for running linux |
| 03:22.18 | dlynes | imagine the ami and agi code you could write with that sucker |
| 03:22.32 | ian6 | no |
| 03:22.33 | ian6 | YOU imagine it. |
| 03:22.34 | ian6 | >:| |
| 03:22.47 | dlynes | You could even control a huge light show on the empire state building using asterisk |
| 03:22.52 | CaT[t3] | i dunno. 212k can buy 20 chunky arsed multicore intels. |
| 03:22.58 | dlynes | forget asterisk as a phone system...think building automation :) |
| 03:24.04 | toresbe | uhm, dual processor? |
| 03:24.13 | dlynes | toresbe: yes |
| 03:24.28 | dlynes | toresbe: minimum configuration |
| 03:24.36 | toresbe | That's very minimum indeed. |
| 03:24.44 | toresbe | We're talking zSeries, right? |
| 03:25.10 | toresbe | I'm impressed they got the price down to a measly 200 000 |
| 03:25.15 | *** join/#asterisk loather-work (n=khudson@internal-nat.djnetworks.net) |
| 03:25.35 | dlynes | toresbe: yeah...z series |
| 03:25.50 | dlynes | toresbe: last year they got one down to 100K |
| 03:25.56 | toresbe | Wow, that's weird. |
| 03:26.00 | dlynes | toresbe: the z10 business class |
| 03:26.24 | dlynes | toresbe: but this is their enterprise server line that they got down to $200K |
| 03:26.24 | toresbe | I want zSeries to be unattainable. It's no fun if they're... attainable. :) |
| 03:26.36 | dlynes | toresbe: they were all in the millions before |
| 03:27.06 | toresbe | Then again, I do stroll around with punch cards in my jacket pocket as note cards, so... |
| 03:27.23 | dlynes | toresbe: well...thye're not much good for much of anything else nowadays |
| 03:27.28 | *** join/#asterisk the_limit (n=the_limi@75-150-44-61-Oregon.hfc.comcastbusiness.net) |
| 03:27.35 | dlynes | toresbe: last ones I seen retired about 25 years ago |
| 03:27.45 | toresbe | oh, you'd be surprised. :) |
| 03:27.56 | dlynes | toresbe: there's some still used? |
| 03:28.01 | toresbe | absolutely |
| 03:28.16 | dlynes | toresbe: and i suppose you're going to tell me that flippies are still used, too? |
| 03:28.18 | toresbe | the 2004 election was a prominent example :P |
| 03:28.34 | lost_soul | so, anyone have any idea how to get pjsua to play a media file or make it ring when a call is incoming |
| 03:28.47 | dlynes | lost_soul: pjsua? |
| 03:29.05 | lost_soul | dlynes: yes, it's a CLI based softphone |
| 03:29.23 | lost_soul | unfortunately atm I don't have the money to get an IP phone |
| 03:29.38 | dlynes | lost_soul: is it compiled code? |
| 03:30.18 | lost_soul | dlynes: I used the package available for openbsd. I could easily remove that and compile it though if you think that would help |
| 03:30.20 | dlynes | lost_soul: and it doesn't do something as basic as ringing, currently? |
| 03:30.43 | dlynes | lost_soul: ....oh...you're on openbsd....does it have sound? |
| 03:30.45 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 03:30.46 | dlynes | ducks. |
| 03:30.46 | lost_soul | dlynes: nope, it shows the call coming in on the CLI, but thats it |
| 03:31.01 | lost_soul | dlynes: lol, yes. Sound works fine |
| 03:31.16 | kam187 | hmm getting delayed voice on asterisk |
| 03:31.21 | kam187 | any general things to check? |
| 03:31.35 | lost_soul | network latency maybe? |
| 03:31.35 | dlynes | kam187: make sure you're not running on your mother's 386 |
| 03:31.41 | lost_soul | lmao |
| 03:31.53 | kam187 | lol |
| 03:31.57 | kam187 | nah i'm not :) |
| 03:32.07 | dlynes | ok...well that's a start, then |
| 03:32.23 | dlynes | what do you mean by 'delayed'? |
| 03:32.42 | dlynes | is there an *echo* *echo* *echo* in the room? |
| 03:32.58 | kam187 | Cpu(s): 0.3%us, 0.0%sy, |
| 03:33.20 | kam187 | i say 1, the other end hears it just under 1 second later |
| 03:33.31 | dlynes | kam187: all the time, or just some of the time? |
| 03:33.36 | kam187 | all the time |
| 03:33.44 | kam187 | well its a new system, but i'm counted to 50 |
| 03:33.49 | kam187 | with the other person counting back to me |
| 03:33.58 | kam187 | its consitantly about 1 second |
| 03:34.13 | kam187 | we're in the same room tho |
| 03:34.15 | dlynes | how are the two systems connected together? |
| 03:34.32 | dlynes | what kind of switch are you using? |
| 03:34.45 | kam187 | laptop running sjphone ---ip cable--> asterisk ---> gsm modem |
| 03:34.51 | kam187 | so its calling another gsm phone |
| 03:35.03 | dlynes | do you think that maybe it might have something to do with the gsm network? |
| 03:35.06 | kam187 | i have my laptop directly into the asterisk with crossover |
| 03:35.10 | lost_soul | I've had that happen before, very hard to diagnose where the problem lies. Actually I found that the "delay" would get better or worse depending on which softphone software I used. |
| 03:35.19 | kam187 | i dont think so, because if i make a call on my mobile there's no delay |
| 03:35.33 | kam187 | hmmm interesting |
| 03:35.40 | dlynes | kam187: have you tried it without a softphone, like lost_soul suggests? |
| 03:35.43 | kam187 | i guess i could do a 'loop back' in the channel driver |
| 03:35.47 | lost_soul | which leads me to believe that with proper equipment(an actual ip phone) that the problem might not be a problem anymore |
| 03:35.49 | kam187 | not yet |
| 03:36.02 | kam187 | its only in one direction btw |
| 03:36.11 | lost_soul | yea, you don't hear it |
| 03:36.12 | lost_soul | they do |
| 03:36.20 | kam187 | asterisk -> gsm modem has 1 second delay |
| 03:36.23 | kam187 | gsm modem to me is fine |
| 03:36.26 | lost_soul | yours likely sounds crystal clear |
| 03:36.40 | dlynes | I've never used a softphone, so I don't know all the issues with them |
| 03:36.49 | kam187 | yeah so what i mean is, if its a gsm delay their voice to me would have a delay too |
| 03:37.04 | *** part/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 03:37.18 | lost_soul | kam187: what softphones are you using? |
| 03:37.22 | kam187 | sjphone |
| 03:37.26 | kam187 | (ulaw) |
| 03:37.49 | dlynes | lost_soul: sjcua and sjphone the same softphone? |
| 03:37.55 | lost_soul | maybe try eyebeam at some point, see if that make it slightly better |
| 03:38.12 | lost_soul | pjsua is completely different |
| 03:38.50 | lost_soul | I used my softphones mostly on windows systems which is why I pointed them to eyebeam, if their on a linux system that makes it a bit more difficult |
| 03:39.33 | lost_soul | I will say pjsua the "delay" is much better than on the windows softphones. Just can't get it to ring or play media so I'm able to know when calls are coming in |
| 03:40.22 | lost_soul | kam187: do you also notice issues when you both try to speak at the same time? |
| 03:40.29 | kam187 | nope |
| 03:40.43 | *** join/#asterisk psilikon (n=joel@221-116.96-97.tampabay.res.rr.com) |
| 03:40.49 | kam187 | although 'same time' is hard to define with a 1 second delay :) |
| 03:41.25 | dlynes | ibm should be releasing debian on those new machines, though |
| 03:41.28 | dlynes | that would kick ass |
| 03:42.12 | lost_soul | kam187: well, to further explain. I've noticed before while testing the various softphones that some seem to not work properly in full duplex mode and if both people speak at the same time, one side is almost like muted. |
| 03:42.35 | dlynes | lost_soul: that could be your crappy sound card |
| 03:42.38 | lost_soul | you can hear them once in awhile, but not all the time |
| 03:42.43 | dlynes | lost_soul: a lot of sound cards are half duplex |
| 03:43.04 | lost_soul | dlynes: only problem was with the softphones |
| 03:43.07 | psilikon | What makes sip extensions show up as ports other than 5060 and 5061? Can you not have multiple extensions share the same ports opened on a firewall? |
| 03:43.17 | dongs | 12:21 < dlynes> lost_soul: a lot of sound cards are half duplex |
| 03:43.26 | dongs | dlynes: youre kidding. maybe in 1995 or something. |
| 03:44.39 | kam187 | interesting |
| 03:44.42 | lost_soul | dlynes: this was with onboard sound, tho that onboard sound has always been quite decent. However I've never put in a pci card and tried with that |
| 03:44.44 | kam187 | its a channel driver problem it seems |
| 03:45.19 | kam187 | its coded to discard voice untill answer |
| 03:45.48 | kam187 | if i take that out, the voice is now 5 seconds delays (5 seconds = time of first voice in -> answer) |
| 03:46.18 | [TK]D-Fender | psilikon: Like what? |
| 03:46.20 | dlynes | dongs: actually...a lot of the winmodems that have built-in sound cards are half duplex sound cards |
| 03:47.16 | lost_soul | :o winmodems |
| 03:47.29 | lost_soul | and you implied I use shitty hardware :p |
| 03:48.06 | psilikon | [TK]D-Fender, I have two ATAs and four extensions behind one router. Does each extension need to be configured with it's own port? Or can multiple extension share the same port? |
| 03:48.15 | dlynes | lost_soul: no...just a shitty OS :) |
| 03:48.20 | lost_soul | :O |
| 03:48.42 | lost_soul | oh stop.. obsd is awesome, just not best choice for a desktop system |
| 03:49.24 | dlynes | lost_soul: frankly...the only bsd i would consider for a desktop would be the mac |
| 03:49.32 | [TK]D-Fender | ~bsd |
| 03:49.33 | infobot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 03:50.00 | lost_soul | [TK]D-Fender: lol |
| 03:50.04 | [TK]D-Fender | psilikon: its because you need to be able to address each of the port independently for INCOMING |
| 03:50.16 | dlynes | [TK]D-Fender: there's also a port (including drivers) for Solaris on the SunSPARC...but I wouldn't recommend it |
| 03:51.02 | lost_soul | dlynes: this is more or less to get more comfy with it.. I used obsd for my router which is also where my asterisk resides. I liked it so threw it on my workstation. |
| 03:51.22 | [TK]D-Fender | dlynes: Preaching to the choir some more? |
| 03:51.22 | p3nguin | Just because you don't use the OS or can't use it does not make it a bad/shitty OS. |
| 03:51.23 | dlynes | lost_soul: ah...it makes sense for your router |
| 03:51.35 | dlynes | [TK]D-Fender: yes |
| 03:51.51 | dlynes | [TK]D-Fender: callweaver is also quite a nightmare on solaris |
| 03:51.54 | dongs | uhhhhh |
| 03:51.56 | dongs | p->stimer->st_interval * 1000 / 2 |
| 03:51.58 | dongs | what is the purpose of /2 |
| 03:52.00 | dongs | in this? |
| 03:52.14 | dlynes | dongs: it's shorthand for 500 |
| 03:52.22 | dongs | no wai? |
| 03:52.33 | dongs | why is it halving the requested timer value on start? |
| 03:52.53 | dongs | do asterisk timers work in 2ms/tick? |
| 03:54.16 | psilikon | [TK]D-Fender, hmmm. I am just trying to figure out why I have two extensions on the *same* lan have no audio (either way). They extension rings though... |
| 03:54.43 | psilikon | Er... the extensions ring |
| 03:56.01 | *** join/#asterisk BugKhaM (n=BugKhaM@125.25.72.189) |
| 03:56.02 | dongs | well, took out /2 for now and see what happens |
| 03:56.26 | [TK]D-Fender | psilikon: First I fail to see models mentioned, and FIREWALLS dumped to be examined |
| 03:57.03 | BugKhaM | I am about to downgrade from asterisk 1.6.1 to 1.6.0. Is there any significant change in the *.conf I should consider? |
| 03:58.05 | dongs | if i dont fix these sesion timers im gonna downgrade to 1.4 where an acutal working patch for it exists |
| 03:59.05 | [TK]D-Fender | BugKhaM: Go read the upgrade.txt for 1.6.1 and see if you did any of that |
| 03:59.27 | psilikon | [TK]D-Fender, Right now I am going to exclude firewalls since I am not going thru any. I am only trying to place a local call between two sip extensions on the same lan and reg'd to the same PBX. What type of debugging could I do? "show sip peers" shows the extension as being on port 1027 for some reason but the status is "OK" |
| 03:59.56 | [TK]D-Fender | psilikon: Says you, and yes, there is a firewall.... |
| 04:00.04 | [TK]D-Fender | psilikon: and you have not NAMED NAMES. |
| 04:00.29 | [TK]D-Fender | And there is no such thing as a "SIP extension" |
| 04:00.57 | dlynes | dongs: ./channels/chan_sip.c:257:#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */ |
| 04:01.04 | dlynes | dongs: perhaps that could explain it? |
| 04:01.14 | dongs | thats different. |
| 04:01.37 | dongs | i'm talking about start_session_timer() |
| 04:01.37 | psilikon | [TK]D-Fender, then what do I call it. Just an extensions? |
| 04:01.49 | dongs | it takes interval in seconds, turns it into ms (*1000) and for some reason halves it. |
| 04:02.06 | [TK]D-Fender | psilikon: I asked for names. Get to it already, I don't want to ask a third time. |
| 04:02.53 | dlynes | dongs: because it needs to retransmit every 500 ms |
| 04:03.08 | dongs | 12:40 < dongs> it takes interval in seconds |
| 04:03.22 | dongs | anyway, im 20 seconds from finding out if its gonna get dropped this time or not |
| 04:03.35 | psilikon | [TK]D-Fender, :) the names of the extensions are 111 and 900. My name is Billy. |
| 04:03.42 | dongs | yep, still dropped. |
| 04:03.43 | [TK]D-Fender | psilikon: .... |
| 04:04.16 | [TK]D-Fender | psilikon: it doesn't seem to be sinking in... |
| 04:04.34 | [TK]D-Fender | psilikon: I'm going to try one last time to be loud and clear. |
| 04:04.58 | BugKhaM | [TK]D-Fender, OK |
| 04:05.16 | [TK]D-Fender | psilikon: what fucking HARDWARE / SOFTWARE represents this "SIP extension". |
| 04:06.04 | [TK]D-Fender | (BOTH actually) |
| 04:06.04 | psilikon | [TK]D-Fender, ohhh. Both extensions are Sipura spa-2001's. |
| 04:06.32 | [TK]D-Fender | psilikon: Good, then ther should only be *1* firewall to contend with |
| 04:07.03 | [TK]D-Fender | psilikon: Go dump it so we can see |
| 04:08.48 | psilikon | [TK]D-Fender, You want to see the iptables on the pbx or you need to see the forwards on my router? |
| 04:09.15 | [TK]D-Fender | psilikon: The call is between * and 2 LOCAL SPA-2001's, correct? |
| 04:09.35 | psilikon | [TK]D-Fender, yep |
| 04:09.49 | [TK]D-Fender | psilikon: Then clearly your router is not in play. Do the math |
| 04:10.34 | psilikon | [TK]D-Fender, Alright bro, what file you need dumped? /etc/iproute2/??? |
| 04:10.52 | [TK]D-Fender | psilikon: iptabes --list |
| 04:11.02 | p3nguin | psilikon: An extension is a numbers, letters, or a combination of the two that you can see in extensions.conf that come after "exten =>" |
| 04:11.09 | p3nguin | iptables -L |
| 04:11.12 | p3nguin | iptables -L -v |
| 04:12.15 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 04:13.08 | psilikon | http://pastie.org/736781 |
| 04:14.43 | [TK]D-Fender | psilikon: iptables --flush |
| 04:14.52 | [TK]D-Fender | psilikon: TRASH IT and retest |
| 04:16.03 | psilikon | But if I disable iptables and I still have the same problem isn't something else wrong? |
| 04:16.12 | psilikon | Ok still the same thing. |
| 04:16.29 | [TK]D-Fender | psilikon: pastebin a failed call with SIP debug enabled |
| 04:18.08 | *** join/#asterisk Newsam (n=a@65.48.179.205) |
| 04:18.17 | Newsam | hi |
| 04:23.22 | *** join/#asterisk lordmortis (n=lordmort@gateway.qsix.com.au) |
| 04:29.08 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 04:30.54 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 04:31.24 | *** join/#asterisk dkirker (n=dkirker@pcp065185pcs.wireless.calpoly.edu) |
| 04:35.20 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 04:36.06 | *** join/#asterisk dkirker (n=dkirker@pcp065185pcs.wireless.calpoly.edu) |
| 04:40.32 | *** join/#asterisk DMeloUK (n=DominicM@127.159.119.70.cfl.res.rr.com) |
| 04:41.19 | *** join/#asterisk dkirker (n=dkirker@pcp065185pcs.wireless.calpoly.edu) |
| 04:41.32 | DMeloUK | trying to build a lean asterisk server for an internal project - do I need dadhi or wanpipe? |
| 04:42.00 | DMeloUK | trying to be as barebones as possible |
| 04:42.32 | CrazyTux | Hey guys I've installed asterisk-addons, and I am attempting to connect however for some reason I am getting connection failed.....for mysql_real_connect |
| 04:42.59 | CrazyTux | on line 301 |
| 04:43.50 | CrazyTux | Wondering if anyone has experienced any similar behavior: Asterisk-addons version: 1.4.10 and asterisk version 1.4.27 |
| 04:45.54 | ChannelZ | DMeloUK: You only need DAHDI for dahdi_dummy if you want to use conferences |
| 04:46.55 | DMeloUK | ChannelZ: so I can skip wanpipe altogether then since I am running SIP/IAX only |
| 04:48.43 | ChannelZ | wanpipe is for sangoma hardware (just as DAHDI is for Digium/clone hardware with the exception of dummy which is for peeps with no hardware) |
| 04:48.58 | DMeloUK | thanks for the clarification |
| 04:49.26 | ChannelZ | well I should say wanpipe is for a lot of things I think but in any case no you dont really need it unless you need it. |
| 04:50.01 | [TK]D-Fender | I can say Wanpipe is for sangoma hardware, and POSSIBLY for *'s benifit therein |
| 04:52.42 | dongs | is SER still alive or what. |
| 04:52.48 | dongs | im not getting anywehre with working session-timers. |
| 04:53.03 | dongs | iptel.org seems dead. |
| 04:56.58 | CrazyTux | dongs: there is a new branch |
| 04:57.14 | CrazyTux | dongs: checkout kamailio or opensips |
| 05:05.43 | dongs | wow, no kidding |
| 05:05.49 | dongs | nice duplication of effort |
| 05:06.26 | dongs | i guess kamailio is what i want. |
| 05:09.53 | dongs | "make future forking undesirable, this harms everybody, affects credibility and business" |
| 05:10.05 | dongs | true words, too bad nobody actually follows through with that. |
| 05:10.17 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 05:10.26 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 05:11.32 | bcrisp | . |
| 05:11.49 | kam187 | anyone use ooh323? |
| 05:12.04 | bcrisp | i disabled it in my build |
| 05:12.15 | kam187 | ah |
| 05:12.21 | kam187 | do u use any h323 at all? |
| 05:12.25 | kam187 | or does anyone else? |
| 05:12.37 | bcrisp | nah |
| 05:12.45 | dongs | i used to use h323 in like, 2000. |
| 05:13.17 | bcrisp | did it work ok? |
| 05:13.38 | dongs | other than taking half a day to compile, yeah. |
| 05:14.00 | bcrisp | it took me 1/2 a day to install SP1 for vs 2005 lol |
| 05:14.07 | dongs | oh my, SER is compiling with -funroll-loops and -O9 |
| 05:14.09 | dongs | it must be real fast. |
| 05:14.51 | dongs | and its hardcoding config to /usr/local... what a joke |
| 05:14.52 | dongs | stopping. |
| 05:15.14 | bcrisp | dongs i think you need a vacation |
| 05:15.23 | toresbe | Yeah, or euthanasia |
| 05:15.29 | toresbe | Do you do anything but complain? |
| 05:15.40 | dongs | toresbe: do you hear me complaining about my windows machines? |
| 05:15.44 | dongs | no because they simply work. |
| 05:15.51 | toresbe | ahahahahaa |
| 05:15.55 | bcrisp | hmm |
| 05:16.04 | dongs | i'm a normal user who just wants shit to work |
| 05:16.15 | toresbe | dongs: You know, you don't get a cookie for not complaining about one subject. |
| 05:16.16 | dongs | instead i spent half a day looking at source of chan_sip because shit doent work. |
| 05:16.19 | bcrisp | then buy a turnkey solution? |
| 05:16.35 | toresbe | dongs: If you were smarter, then maybe you'd use less time. |
| 05:16.36 | dongs | bcrisp: it wouldnt be profitable for them to sell it to me. |
| 05:17.02 | dongs | because it wont work any better than unsupported shit, and instead they will ahve to waste time (jsut like I'm doing now) making it work for me. |
| 05:17.09 | toresbe | or even less insufferable would help |
| 05:17.21 | bcrisp | roll your own? |
| 05:18.34 | bcrisp | anyways |
| 05:19.13 | dongs | i think you misspelled 'rofl your own' |
| 05:19.30 | bcrisp | hmm |
| 05:19.48 | bcrisp | if you hate it and it doesn't work and you don't want to buy one, why not write your own? |
| 05:20.32 | [TK]D-Fender | dongs: if its so bad... what aren't all of us freaking out just like you? |
| 05:20.49 | *** part/#asterisk CrazyTux (n=brandon@70-138-119-70.lightspeed.hstntx.sbcglobal.net) |
| 05:20.52 | dongs | what? |
| 05:20.59 | dongs | sorry, missed the point in that last sentence. |
| 05:21.19 | [TK]D-Fender | [00:16]<dongs>instead i spent half a day looking at source of chan_sip because shit doent work. <---- |
| 05:21.31 | [TK]D-Fender | dongs: Seems to work for jsut about everybody else |
| 05:21.45 | dongs | [TK]D-Fender: 100% of chan_sip users do not have session timers. |
| 05:22.05 | [TK]D-Fender | dongs: Only you seem to have found a huge issue with this "fact" |
| 05:22.07 | dongs | you might remember i've had these issues for years. |
| 05:22.35 | [TK]D-Fender | dongs: "You" being the operative word. So what are you doing that it poses an issue? |
| 05:22.51 | dongs | i.. use session timers. |
| 05:22.54 | toresbe | here's the fun part, you may well be right, but nobody cares 'cause you act like an asshole. |
| 05:22.55 | dongs | which don't properly work. |
| 05:23.13 | toresbe | welcome to "interacting with other people 101" |
| 05:23.14 | bcrisp | whenever i have "issues for years" i typically sell it, divorce it, or trade in for something else |
| 05:24.01 | [TK]D-Fender | dongs: Feel free to submit a patch, port the one you've found, or put something else in between. |
| 05:24.29 | [TK]D-Fender | dongs: But ranting certainly isn't the solution. |
| 05:26.05 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:26.49 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 05:26.56 | DMeloUK | any reason to go with 1.6.1.11 over 1.6.0.19 ? |
| 05:27.17 | Corydon76-dig | Feature set |
| 05:27.24 | [TK]D-Fender | DMeloUK: For something that 1.6.1 offers that 1.6.0 doesn't |
| 05:29.01 | DMeloUK | lol - I guess I deserved that :) |
| 05:34.44 | dongs | for what reason are asterisk logfiles including ansi color codes in them... lol. |
| 05:38.29 | *** join/#asterisk kerx (n=kerx__@76-240-161-68.lightspeed.irvnca.sbcglobal.net) |
| 05:40.07 | *** join/#asterisk dkirker (n=dkirker@pcp065185pcs.wireless.calpoly.edu) |
| 05:47.09 | *** join/#asterisk geneticx (n=geneticx@adsl-144-252-204.mia.bellsouth.net) |
| 05:51.15 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 05:58.22 | dongs | the logic in proc_session_timer() makes no sense. |
| 05:58.35 | dongs | it will always erorr out unless its called during outgoing call. |
| 05:58.47 | dongs | because only that time res is set to 1, and reinvite is sent to refresh the timer. |
| 05:58.56 | dongs | its just a giant dummy hunk of shit for incoming calls. |
| 05:59.17 | dongs | does anyone QA this shit for more than 5 seconds? |
| 05:59.25 | dongs | or do they just go "oh, i think it works, fuck'it" |
| 06:01.55 | ChannelZ | Hey ahole, why don't you write something better? |
| 06:02.25 | Kobaz | dongs: write unit tests |
| 06:02.29 | dongs | hey ahole, by version 1.6.x a normal user would expect release-quality software. |
| 06:02.39 | ChannelZ | And you paid how much for this? |
| 06:02.40 | Kobaz | dongs: that's the only way to make sure stuff works right, and doesn't regress in new versions |
| 06:02.45 | dongs | Kobaz: go back to writing tcl scripts. |
| 06:02.49 | Kobaz | haha |
| 06:02.49 | dongs | (p.s. hi) |
| 06:02.52 | Kobaz | :) |
| 06:02.59 | Kobaz | i write asterisk modules these days |
| 06:03.06 | Kobaz | and fix asterisk bugs |
| 06:03.16 | dongs | you can start being awesome by fixing this one. |
| 06:03.25 | Kobaz | i don't know anything about the sip communication |
| 06:03.37 | Kobaz | i've just fixed some random tidbits in some applications |
| 06:03.39 | dongs | me neither, and i've already found 2 goatse-sized gaping holes in it. |
| 06:03.50 | Kobaz | like uninitalized variables and crap |
| 06:04.03 | Kobaz | and added some messaged to events where they were missing |
| 06:04.13 | Kobaz | wrote two new apps, and a function |
| 06:04.27 | Kobaz | once i port everything to trunk i'm submitting it all |
| 06:04.39 | Kobaz | goatse-sized |
| 06:04.41 | Kobaz | that's pretty big |
| 06:04.54 | ChannelZ | One of these people is a productive citizen. The other is not. Can anyone guess which is which? |
| 06:05.12 | dongs | you're the non-productive one. |
| 06:05.23 | dongs | because you offer no quality input to either asterisk or this conversaion. |
| 06:05.25 | Kobaz | hehe |
| 06:05.36 | ChannelZ | Well I was able to read UPGRADE.txt and answer one of your questions, so there's that. |
| 06:05.50 | Kobaz | mmm, 5 hours left on this disk mirror |
| 06:05.55 | Kobaz | i should nap |
| 06:07.04 | Kobaz | i find asterisk to be an awsesomely powerful platform, and very flexable |
| 06:07.17 | Kobaz | but, it sucks when it randomly crashes when it's been running for 2 months straight |
| 06:07.22 | Kobaz | doing the same thing, day in and day out |
| 06:08.35 | dongs | maybe thats why they compile it wiht -g3 by default |
| 06:08.41 | Kobaz | dongs: i thought my eggdrop tcl was pretty good... i started that project when I was like, 12 |
| 06:09.01 | Kobaz | you also need to compile with DONT_OPTIMIZE |
| 06:09.02 | dongs | i wasnt serious about that relax. |
| 06:09.06 | Kobaz | so you can get the full backtrace |
| 06:09.12 | Kobaz | dong: hehe, i know... but i'm just saying |
| 06:09.19 | Kobaz | i think the collection turned out pretty good |
| 06:09.31 | Kobaz | the last bug report i had was like 2 years ago |
| 06:10.03 | Kobaz | the fact of the matter is |
| 06:10.09 | Kobaz | most people, write really shitty code |
| 06:10.28 | tzafrir__laptop | DONT_OPTIMIZE is not related to a "full backtrace" |
| 06:10.30 | Kobaz | it takes a lot of time, analysis, and refactoring... to make something very good |
| 06:11.04 | tzafrir__laptop | You build with DONT_OPTIMIZE if you want to have very non-optimized code that has what you actually wrote |
| 06:11.11 | Kobaz | tzafrir__laptop: stuff get's optimized out |
| 06:11.20 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 06:11.24 | Kobaz | and it's harder to debug |
| 06:11.41 | tzafrir__laptop | You should use -g regardless - it has almost no run-time over |
| 06:11.50 | Kobaz | yeah it's just symbols |
| 06:12.06 | tzafrir__laptop | Yeah, but I'm not so fond of wasting my production system's CPU time |
| 06:12.22 | Kobaz | well |
| 06:12.34 | Kobaz | i would rather be able to fully debug spurious crashes |
| 06:12.59 | Kobaz | none of my systems have a high enough load that -O3 would really make a difference |
| 06:14.08 | Kobaz | i almost want to run valgrind full time on production systems |
| 06:14.19 | Kobaz | but that REALLY puts a hit on the cpu, and it's just not doable |
| 06:14.56 | Kobaz | i think the safest thing is to have about 10 asterisk's running on the same box |
| 06:15.00 | Kobaz | and load balance between them all |
| 06:15.18 | Kobaz | so when a channel hangup triggers a double free, or a heap corruption, it'll only take out a few calls |
| 06:16.10 | dongs | alternatively, you could just write software that doesnt crash. |
| 06:16.20 | Kobaz | tell that to the asterisk devs :P |
| 06:16.49 | Kobaz | i've seriously considered moving to freeswitch, or something |
| 06:16.54 | Kobaz | but they don't do unit testing either |
| 06:17.07 | Kobaz | and i do have a lot invested in asterisk-specific design |
| 06:17.30 | dongs | i just want something that works |
| 06:17.32 | Kobaz | so.. out project for first-second quarter next year, is writing an asterisk test suite |
| 06:17.51 | Kobaz | so we can know for certain, when we upgrade, that stuff still works |
| 06:18.21 | Kobaz | if perl can do it, so can asterisk |
| 06:18.32 | Kobaz | the community momentum just needs to be built up |
| 06:18.39 | Kobaz | every damn perl module in cpan has unit tests |
| 06:19.14 | dongs | hey, freeswitch does sip session timers. |
| 06:19.24 | dongs | maybe its time to free"switch". |
| 06:19.59 | Kobaz | hehe |
| 06:20.08 | dongs | though i just added some hack in that session timer proc and gonna see if it will still get me hung up or not |
| 06:20.20 | Kobaz | so what's the issue you're having? |
| 06:20.39 | dongs | provider drops me due to improper session timer implementation after 4min20sec of call |
| 06:20.47 | Kobaz | ah |
| 06:20.49 | Kobaz | wiggity |
| 06:20.52 | Kobaz | i've never had that |
| 06:20.52 | dongs | tehy wanna refresh , this stuff doesnt, they drop call |
| 06:21.02 | dongs | you dont, theres like 2 providers that actually enforce session timers. |
| 06:21.05 | Kobaz | hehe |
| 06:21.31 | dongs | nope. |
| 06:21.34 | dongs | dropped. |
| 06:21.39 | dongs | well, freeswitch it is. |
| 06:21.42 | Kobaz | heh |
| 06:21.50 | Kobaz | everything is all xml |
| 06:22.18 | dongs | 31megs :| |
| 06:22.29 | Kobaz | i had some bad experiences with xml a few years ago, i think i still have ptsd |
| 06:23.21 | Kobaz | it wasn't xml specifically |
| 06:23.35 | Kobaz | retarted idiot i worked for, was using xml over udp |
| 06:23.49 | Kobaz | and writing his own special multi part udp packet handling and shit |
| 06:23.58 | dongs | asterisk still doesnt do 16khz audio? or are these guys trolling |
| 06:24.01 | Kobaz | i'm like god damnit, tcp does all that for you |
| 06:24.10 | Kobaz | no, no 16khz audio |
| 06:24.18 | drmessano | [01:19] <dongs> hey, freeswitch does sip session timers. [01:19] <dongs> maybe its time to free"switch". <--- Don't let the door hit your ass on the way out! |
| 06:25.48 | Kobaz | heh |
| 06:25.57 | Kobaz | i complain, but i still use asterisk |
| 06:26.02 | *** join/#asterisk ethicx (n=geneticx@adsl-144-252-204.mia.bellsouth.net) |
| 06:26.09 | Kobaz | i want to see it, and help it get better |
| 06:26.33 | [TK]D-Fender | waits for the head to align between the door & frame and gives it a hearty push |
| 06:27.34 | [TK]D-Fender | dongs: Oh.. and you have a patch for 1.4 Why aren't you using it? |
| 06:27.35 | drmessano | It takes a special kind of unhelpful douchebag to complain for years about something not being implemented properly in asterisk to either not fix it himself, or give a compelling reason anyone should spend real time on it to help 1 effin person |
| 06:27.49 | Kobaz | heh |
| 06:28.09 | dongs | [TK]D-Fender: no idea. i figured in 2009 i should use something from 2009, instead of something from 2004 |
| 06:28.10 | [TK]D-Fender | drmessano: Oh he HAS a patch... for 1.4... which I suppose he could either use, adapt to a newer version himself, or contract someone else to. |
| 06:28.31 | dongs | adapting it to newer version wouldnt work as 1.4 doesnt support session timers in any shape and the patch is just a hack. |
| 06:28.48 | [TK]D-Fender | dongs: Who cares about labels if it works... |
| 06:28.51 | dongs | but 1.6 "does" support it, brokenly. |
| 06:29.12 | drmessano | [TK]D-Fender: At least he made a half assed attempt at not getting it working. Sounds like he really needs this feature and doesnt just need to bitch |
| 06:29.23 | Kobaz | rentacoder.com |
| 06:29.59 | [TK]D-Fender | drmessano: No... "bitching" came bundled with the package :) |
| 06:31.43 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 06:33.36 | drmessano | He compiled with BE_APART_OFTHE_PROBLEM ? |
| 06:35.23 | *** join/#asterisk ticoit (n=ticoit@201.191.151.147) |
| 06:35.45 | dongs | haha configure is still running for this stuff |
| 06:35.47 | dongs | im not sure i wanna continue. |
| 06:35.56 | dongs | i saw it compiling libtiff and other utterly irrelevant stuff |
| 06:35.59 | dongs | waht happened to shit that just works |
| 06:36.23 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 06:36.26 | dongs | why does a SIP server need libtiff? |
| 06:36.32 | dongs | question of the day. |
| 06:36.43 | drmessano | Fax, you dong |
| 06:36.50 | Superbartt | dongs because a SIP server kan do a lot more than just voice ;) |
| 06:36.51 | drmessano | Try unchecking something |
| 06:38.02 | tzafrir__laptop | TIFF is the common format for faxes |
| 06:38.02 | drmessano | Try being smug about something that doesnt make you look like an incredibly ignorant newb :) |
| 06:38.12 | drmessano | Just sayin :0 |
| 06:38.22 | drmessano | fake pretentious smiley :) |
| 06:38.45 | drmessano | reaches for the extra large TrollSmasher 2000(C) |
| 06:39.05 | [TK]D-Fender | Or as a format for taking a still-pic from a video media stream <- |
| 06:39.39 | drmessano | It's also used to seed the SIP session timers |
| 06:40.03 | [TK]D-Fender | Yes... it transmits pictures of clocks to reference it by! |
| 06:40.18 | Superbartt | lol |
| 06:43.12 | drmessano | lol |
| 06:44.02 | drmessano | Every 20 minutes it sends a pick of exclaimated "DONG!" |
| 06:44.07 | drmessano | pic |
| 06:44.28 | drmessano | In tribute to the dongmaster himself |
| 06:45.16 | drmessano | You want something to bitch about.. Windows 7 doesnt support my USB <> SERIAL adapter out of the box |
| 06:47.46 | [TK]D-Fender | drmessano: Plugging them in unprotected like that risks getting a virus. You need to keep in in the box ;) |
| 06:48.08 | [TK]D-Fender | drmessano: dong-le :) |
| 06:48.26 | drmessano | lol |
| 06:48.34 | [TK]D-Fender | ok, bedtime... later all |
| 06:48.51 | drmessano | I found a Vista driver for it, which BSODs it every 300 or so lines of data |
| 06:49.14 | drmessano | Which worked well enough to get a shitty old PIX 501 box repurposed for Radiothon |
| 06:49.38 | drmessano | I really should sell it on ebay and buy a half dozen WRT54Gs I can actually USE |
| 06:49.41 | *** join/#asterisk Tim_Toady (n=moi@77.49.57.163) |
| 06:50.41 | dongs | you have to try real hard to buy a usb serial adapter that isnt based on FTDI chip or doesnt use USB comm device class,, so you've got nobody to blame but yourself for it. |
| 06:51.22 | dongs | cool. freeswitch compile failed because I'm not gonna install "ODBC" |
| 06:51.27 | dongs | trashing that idea. |
| 06:51.40 | *** join/#asterisk |Cybex| (n=John@212.178.82.21) |
| 06:51.58 | dongs | ill make sure to let htem know this |
| 06:53.21 | tzafrir__laptop | dongs, them: #freeswitch |
| 06:54.04 | dongs | yep, already there. |
| 06:54.19 | drmessano | Be sure to compliment them on their working session timers |
| 06:54.32 | dongs | i am sure i will once it actually works. |
| 06:54.36 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 06:54.43 | eppigy | YEAH |
| 07:03.37 | bcrisp | dude wow thats like 7 hrs of solid moaning.. nice! |
| 07:04.21 | eppigy | say what |
| 07:07.07 | *** part/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 07:07.57 | kam187 | hmm when i try to assign a hangup cause to pvt->owner->hangupcause asterisk segfaults |
| 07:07.59 | kam187 | wtf |
| 07:18.20 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:21.38 | *** join/#asterisk xrmx__ (n=rm@host143-252-dynamic.8-87-r.retail.telecomitalia.it) |
| 07:22.27 | *** join/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 07:30.22 | *** join/#asterisk cpoulson (n=root@pdpc/supporter/active/cpoulson) |
| 07:45.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 07:47.43 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 07:50.21 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:50.26 | *** join/#asterisk dlu_ (n=quassel@83.125.8.7) |
| 07:50.34 | dlu_ | hi all |
| 07:51.06 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:51.18 | dlu_ | anyone here can help answer me a sip related question about asterisk 1.6? i have tried to find a answer in the changelogs but dintd find any |
| 07:51.52 | dlu_ | i habe asterisk as registrar, kamailio as outbound router and mediagateways to terminate the calls |
| 07:53.09 | dlu_ | if i do an outbound with asterisk 1.6 the call go out abut after the okay msg (off hook) i release the call and the asterisk didnt send the bye to the kamailio it send it direct to the mediagateway |
| 07:53.26 | dlu_ | that is an behaviour that i dont want to practice |
| 07:53.48 | dlu_ | is there a switch to let kamailio in the siganllingpath? |
| 07:55.47 | dongs | maybe canreinvite=no or something, but i doubt its related. |
| 07:55.52 | dongs | just ignore me, i have no idea |
| 07:56.03 | dlu_ | :-) |
| 07:56.45 | dlu_ | btw canreinvite is set to no |
| 07:57.00 | dlu_ | its the same config i use in asterisk 1.4 |
| 07:57.21 | dlu_ | that is really sucking me off |
| 07:57.34 | ChannelZ | lordy |
| 07:58.14 | dlu_ | i dont be a lamo on asterisk because i use it since 4 year and have many experiance but what asterisk still do is very crazy |
| 07:59.49 | dlu_ | i looked in the okay message and ther is still only a via with the ip address of the medagate and yes the sdp have the rtp stream ip. but i dont guess asterisk expecting the "new" sip endpoint at the rtp address |
| 08:00.23 | dlu_ | or still it does? |
| 08:02.14 | dongs | just go back to 1.4, im gonna do that shortly as well |
| 08:02.50 | dlu_ | that cant be the solution |
| 08:03.37 | dlu_ | these things must be discussed so i will open a bug :-) |
| 08:03.52 | *** join/#asterisk mintos (n=mvaliyav@117.206.17.28) |
| 08:09.53 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
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| 08:32.53 | drmessano | dlu_: Ignore dongs, he's a troll |
| 08:33.36 | drmessano | dlu_: 1.6.x is great.. glad you're opening a bug and not being a dong about it |
| 08:34.19 | dlu_ | okay oepend a bug about it |
| 08:34.44 | jblack | dongs is still here? |
| 08:35.00 | jblack | that twit has been here for hours. |
| 08:37.27 | ChannelZ | his freeswitch defection lasted all of about 3 minutes (unfortunately) |
| 08:38.05 | ChannelZ | finally installed 1.6.1 at home |
| 08:42.49 | *** join/#asterisk creativx (n=creadure@82.134.19.197) |
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| 08:53.35 | dongs | yea |
| 08:53.51 | dongs | jblack: nice way to treat your non-paying customers |
| 08:55.34 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
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| 09:03.50 | reptiles | wonders if he can get some insight into a DTMF issue he is having. |
| 09:03.59 | reptiles | google is being quite useless today. |
| 09:12.18 | ChannelZ | and the issue is..? |
| 09:18.14 | reptiles | sorry, coffee. |
| 09:18.36 | reptiles | i have 1.4 running on freebsd, and for the most part things are working nicely. |
| 09:18.51 | reptiles | i have a little ATA with some handsets attached, using SIP to connect. |
| 09:19.03 | reptiles | i have a service provider giving me trunks via IAX |
| 09:19.16 | reptiles | and i have a local BRI, which is connected using ISDN4BSD and chan_capi |
| 09:19.46 | reptiles | if i call an IVR system from the handset through the IAX trunks, i can interact with things fine. |
| 09:20.00 | reptiles | however, if i call a local system through the ISDN, it does not work. |
| 09:20.19 | reptiles | if i call from the PSTN, to the ISDN to the handset, i can hear DTMF being send. |
| 09:20.46 | reptiles | if i call from the handset, out of the ISDN to a PSTN line, all i hear is chirps. |
| 09:21.06 | reptiles | so, DTMF incoming on the ISDN is working goodly. |
| 09:21.22 | reptiles | DTMF outgoing on the ISDN is not working usefully. |
| 09:21.25 | reptiles | any ideas? |
| 09:22.46 | ChannelZ | not I, sorry, I don't have any first-hand experience with ISDN |
| 09:23.10 | reptiles | kk |
| 09:23.12 | reptiles | thanx |
| 09:23.54 | *** join/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 09:29.45 | ChannelZ | perhaps a wrong signalling setup in your chan_capi.conf |
| 09:32.17 | dlu_ | correction veriton kill my record-route so the asterisk do all okay |
| 09:32.39 | ChannelZ | well that's good. sort of :) |
| 09:32.39 | dlu_ | correction verizon kill my record-route so the asterisk do all right i guess :S |
| 09:33.14 | dlu_ | iv i get a reply from vzb i will close my bug :S |
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| 09:55.34 | danj1980 | Hello, can anyone give me some advice on Cisco 7940g phones and the licensing involved. |
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| 11:08.52 | Newsam | is anyone here? |
| 11:09.29 | kaldemar | ~ask |
| 11:09.30 | infobot | 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. |
| 11:11.03 | Newsam | i am not familiar with asterisk but i need to set up a phone system with an automated voice menu telling you extensions that are available....the extensions actually go back out to my cell phone and the cell phone of others....is that possible? |
| 11:11.58 | *** join/#asterisk vally (i=vally@ip-92-50-113-160.unitymediagroup.de) |
| 11:12.24 | kaldemar | yes it is, not the simplest task though. |
| 11:13.10 | Newsam | can i do it with one phone line? |
| 11:13.43 | kaldemar | number of phone lines are irrelevant in that. |
| 11:14.57 | kaldemar | but you obviously need to have more lines (in the analog line sense) if you plan to have multiple simultaneous calls. |
| 11:15.04 | kaldemar | this is a good read for you: |
| 11:15.06 | kaldemar | ~thebook |
| 11:15.07 | infobot | well, 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 |
| 11:16.24 | Newsam | hm |
| 11:17.02 | Newsam | so when a caller dials my extension 123 and it goes to my cell phone 2692190 my pbx is still handling that call? |
| 11:17.10 | Newsam | or has it left the system all together? |
| 11:17.11 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 11:21.04 | kaldemar | your pbx is still handling the call |
| 11:21.28 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 11:27.49 | Newsam | ok kaldemar thanks for the links, i will try to read them shortly |
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| 11:44.35 | dongs | well looks like time to downgrade to 1.4 |
| 11:45.52 | TommyBotten | ? |
| 11:47.52 | drmessano | Thought you were doing that 3 1/2 hours ago |
| 11:48.00 | drmessano | Run out of places to bitch? |
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| 11:54.04 | *** join/#asterisk gauravkumar (n=chatzill@114.143.185.90) |
| 11:55.03 | gauravkumar | I finally got the asterisk gui working.. is there any place where i can get some documentation about how to use the digium/asterisk gui? Any help will be appreciated. |
| 11:55.48 | dongs | no, you're supposed to read the source. |
| 11:57.14 | kaldemar | gauravkumar: #asterisk-gui |
| 12:00.07 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 12:02.01 | gauravkumar | thanks, kaldemar |
| 12:10.28 | *** join/#asterisk guyvdb_ (n=guy@dsl-240-182-94.telkomadsl.co.za) |
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| 12:22.58 | gauravkumar | @dongs WHAT??? |
| 12:24.24 | guyvdb_ | hi, I have been having trouble dialing out on DAHDI. I am now getting "No translator path exists for channel type DAHDI" followed by "Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available)" in the CLI when I try dial out via DAHDI |
| 12:28.39 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 12:28.45 | DelphiWorld | hi dear all |
| 12:28.45 | DelphiWorld | please: |
| 12:28.45 | DelphiWorld | need someone from paris or anything in france |
| 12:40.23 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 13:13.17 | ManxPower-work | ~answers |
| 13:13.18 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:15.11 | tzafrir__laptop | is impressed. Two people answered that spam message, but both bothered to properly trim out the spam |
| 13:17.42 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
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| 14:03.26 | *** join/#asterisk Cuz (n=plastik@mail.gradeatechs.com) |
| 14:04.27 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:04.32 | Cuz | hey guys, does anyone in here know if i can virtualize asterisk? I mean, I know it's easilly done, but i guess my real question is, does anyone know if any hypervisor (hyper-v, esx, xen) supports my Sangoma A101D card in a virtualized environment...? :P |
| 14:05.01 | Cuz | i don't even know where to ask, so i thought i'd start here |
| 14:05.27 | Chainsaw | Cuz: It is unlikely to work very well because telephony is a timing-sensitive workload. |
| 14:05.40 | Cuz | good point |
| 14:09.16 | mcargile | If you are recording a channel using monitor with DTMF tones being sent via rfc2833, will those tones show up in the recording? |
| 14:12.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:15.27 | Katty | brr. |
| 14:17.40 | Chainsaw | Hi Katty. |
| 14:19.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.31 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 14:20.12 | Katty | hugs Chainsaw |
| 14:20.25 | Chainsaw | :) |
| 14:22.00 | Katty | scowls |
| 14:22.11 | Katty | it's horribly irritating when someone opens my office door and just stands there talking to someone else |
| 14:22.44 | Katty | if youa re going to come in, come in, otherwise shut my door! |
| 14:22.46 | Katty | grumps. |
| 14:23.41 | Chainsaw | Yes, it's an office, not a meeting room. |
| 14:24.01 | Chainsaw | (It's open plan here, but you get the same problem with hovering chatterboxes) |
| 14:24.05 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:24.05 | Chris-NB | hi |
| 14:24.11 | Chainsaw | Hello Chris. |
| 14:24.13 | Chris-NB | anyone trying SRTP with Asterisk? |
| 14:24.21 | Chainsaw | Plain boring RTP here, sorry. |
| 14:25.49 | Katty | at this point i'm barely verticle, much less thinking about asterisk |
| 14:25.51 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 14:26.36 | Chris-NB | *damned |
| 14:27.02 | Chris-NB | after three years of not thinking about srtp I'm back on it |
| 14:27.20 | *** join/#asterisk RypPn2 (n=Gloop@rossgroup2.demon.co.uk) |
| 14:27.30 | Chris-NB | but it's almost as poor as three years ago |
| 14:27.31 | Chris-NB | : / |
| 14:27.47 | Katty | that should probably be a sign for you |
| 14:31.44 | Cuz | they gave me my own office where i'm at |
| 14:31.56 | Cuz | people were sick of my moods, so they locked me in the basement :P |
| 14:32.24 | Chris-NB | Katty, sry, I'm not the one for coding : ) |
| 14:32.25 | Cuz | its technically the stock room, but hey its got a door, desk, computer and i'm the only one with a key :P |
| 14:32.27 | *** join/#asterisk Akiraa (n=Akira@79.112.38.41) |
| 14:32.33 | Katty | i think computer people are simply grumpy when asked to deal with People |
| 14:34.25 | ManxPower-work | I HATE PEOPLE! |
| 14:34.34 | Gido-E | ManxPower-work why are you online? |
| 14:34.43 | Katty | hugs ManxPower-work |
| 14:34.45 | ManxPower-work | Gido-E: so I don't have to deal with people. |
| 14:34.50 | Katty | Gido-E: isn't that kinda obvious? |
| 14:34.55 | eppigy | GOOD MONRING |
| 14:35.03 | jaytee | TRABAJO! |
| 14:35.08 | eppigy | SI ME GUSTA |
| 14:35.13 | Katty | hi dave |
| 14:35.18 | Katty | and mister jaytee |
| 14:35.22 | Katty | hugs eppigy and jaytee |
| 14:35.40 | eppigy | blushes |
| 14:36.08 | Katty | ^_- |
| 14:36.17 | ManxPower-work | gets Mad Squirrel Disease from Katty |
| 14:36.24 | Katty | muwahahaah |
| 14:37.59 | Katty | i think i will stream WoW this morning. |
| 14:38.09 | Katty | i found this Fun New Program that lets you stream your....screen. |
| 14:38.12 | eppigy | it is so cold |
| 14:38.38 | Chainsaw | Katty: Ah, no more webcam-pointed-at-laptop. Cool. |
| 14:39.01 | Katty | aye. no point (= |
| 14:39.11 | eppigy | lol |
| 14:39.18 | eppigy | 26 degrees :< |
| 14:39.26 | jaytee | according to my little weather applet it's 12F outside |
| 14:39.26 | eppigy | why for |
| 14:39.27 | Chainsaw | C or F? |
| 14:39.30 | eppigy | f |
| 14:39.30 | Katty | yikes. with or without windchill? |
| 14:39.37 | eppigy | with windchill |
| 14:39.39 | Katty | checks blackberry |
| 14:39.43 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:39.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:39.44 | Chainsaw | Ah yes, -3C is a bit cold :/ |
| 14:39.47 | jaytee | without windchill |
| 14:39.48 | Cuz | -15C here |
| 14:40.00 | eppigy | D: |
| 14:40.05 | Katty | Cuz: whare are you so i can avoid it like the plague |
| 14:40.08 | Cuz | its -39C in iqaluit, nunavut :) |
| 14:40.10 | Chainsaw | 48.0 F / 8.9 C here. |
| 14:40.11 | Katty | oh damn. 3F here with windchill |
| 14:40.15 | Katty | 16F otherwise. |
| 14:40.24 | Katty | wind: WNW at 12mph |
| 14:40.31 | jaytee | with windchill it feels like -8F |
| 14:40.43 | Cuz | Katty: canada's capital :D |
| 14:41.07 | Katty | jaytee: eeek |
| 14:41.07 | jaytee | that's Ottawa? |
| 14:41.13 | Cuz | indeed |
| 14:41.26 | ManxPower-work | 25F here in Huntsville, AL |
| 14:41.28 | Cuz | i'm glad you didn't say montreal |
| 14:41.50 | [TK]D-Fender | Cuz: Why not? Not entirely inaccurate |
| 14:42.10 | Cuz | um, yes it would be :P |
| 14:42.21 | [TK]D-Fender | Cuz: No, it wouldn't |
| 14:42.38 | Cuz | today, yes it would be |
| 14:42.42 | jaytee | "tastes great!" "less filling!" |
| 14:42.43 | Cuz | lol |
| 14:42.44 | [TK]D-Fender | Cuz: Montreal WAS the first capital of Canada. Feel free to take a history course :) |
| 14:42.47 | Cuz | WAS |
| 14:42.52 | Cuz | why i said today it would be |
| 14:42.53 | Cuz | :P |
| 14:43.08 | jaytee | now, now! let's not dwell |
| 14:43.11 | [TK]D-Fender | Cuz: Hence not "entirely inaccurate". Not my fault you can't think forth-dimensionally. |
| 14:43.11 | Katty | ManxPower-work: is that without the windchill? |
| 14:43.26 | Cuz | not enough qbits in my head i guess |
| 14:43.45 | [TK]D-Fender | hordes his pre-release copy of res_fluxcapacitor.so |
| 14:44.16 | Cuz | you're one of those scary-creepy-ish type nerds, aren't you? |
| 14:44.28 | Katty | no. he's just...obnoxious |
| 14:44.35 | ManxPower-work | Katty: without |
| 14:44.42 | Katty | ManxPower-work: what is it with? |
| 14:44.51 | ManxPower-work | Katty: 17F |
| 14:44.59 | Katty | :< |
| 14:45.03 | Katty | brr |
| 14:45.31 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:45.31 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:45.48 | eppigy | i stopped smoking today |
| 14:45.54 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:46.11 | Katty | did you get the ecigs |
| 14:46.29 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 14:46.37 | eppigy | Katty: negative |
| 14:46.41 | eppigy | I am just quitting |
| 14:46.41 | Zeeek | where is he? |
| 14:46.49 | ManxPower-work | moves away from eppigy |
| 14:46.55 | eppigy | I am fine |
| 14:47.02 | eppigy | chantix works really well |
| 14:47.15 | eppigy | of course talk to me again in 4 hours |
| 14:47.19 | eppigy | and i might be ragin |
| 14:47.58 | Zeeek | anyone here familiar with Voxbone? |
| 14:49.44 | Zeeek | <crickets> |
| 14:49.49 | Zeeek | where is she? |
| 14:51.28 | Zeeek | Where is Katty |
| 14:52.03 | Zeeek | puts out a bowl of carob soy milk and hides |
| 14:57.10 | Katty | eyes bowl |
| 14:57.58 | Zeeek | adds more carob deliciousness |
| 15:02.45 | Katty | give me a min (= |
| 15:04.07 | Zeeek | has the conternts analized and certified environtmentally friendly and produced by free roaming carob trees |
| 15:06.30 | [TK]D-Fender | Yes..t he contents will be "analized" in a rather short period of time.... |
| 15:07.35 | eppigy | lol |
| 15:07.45 | Zeeek | about 12 hours after consumpyion, I'd guess |
| 15:07.49 | eppigy | i'll take over doctor |
| 15:08.02 | Zeeek | hey, where you going with that thing in your hand? |
| 15:08.20 | [TK]D-Fender | Zeeek: You're going to feel a little prick ;) |
| 15:08.35 | Zeeek | I know several of those on Twitter already |
| 15:08.58 | Zeeek | wonders if Canadians say "prick" in that way |
| 15:09.18 | Zeeek | ~prick |
| 15:09.30 | Zeeek | ~douchebag |
| 15:09.59 | Zeeek | ~google prick |
| 15:10.21 | Zeeek | everything is broken here |
| 15:11.24 | Katty | hugs Zeeek |
| 15:11.35 | Katty | you're here early |
| 15:11.42 | Zeeek | 24 hours early |
| 15:11.44 | Katty | what's the occasion? |
| 15:12.16 | *** join/#asterisk ReeceSavage (n=Reece@vyatta.kingballow.com) |
| 15:12.20 | ReeceSavage | I have 2 Rhino cards for sale if anyone needs one. They are both Best Offer. I have a R2T1-EC and a R24FXX-EC with 12 dual FXS modules. Both have never been used more than a few times for testing purposes. Both cards work fine and are guaranteed not to be DOA. |
| 15:12.39 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:12.39 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 15:13.14 | x86 | ReeceSavage: generally it is not acceptible to spam here |
| 15:13.25 | [TK]D-Fender | ReeceSavage: Ebay <- |
| 15:13.59 | ReeceSavage | how was I spaming? I listed it once |
| 15:14.19 | ReeceSavage | Hardly see stuff on Ebay. Not sure they would even sell |
| 15:14.28 | ReeceSavage | Rhino stuff* |
| 15:15.12 | Katty | jameswf works for rhino |
| 15:15.16 | Katty | but he's not here right now |
| 15:16.10 | *** part/#asterisk ReeceSavage (n=Reece@vyatta.kingballow.com) |
| 15:18.29 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 15:19.40 | Zeeek | Katty: occasion? I don need no stinkin' occasion |
| 15:19.55 | Katty | k |
| 15:20.34 | Zeeek | ~seen JerJer |
| 15:20.36 | infobot | jerjer <n=PhatJ@asterisk/original-h323-guy/JerJer> was last seen on IRC in channel #asterisk, 37d 19h 27m 35s ago, saying: 'then tried to blame it on QoS'. |
| 15:20.46 | Zeeek | heh |
| 15:22.57 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 15:23.11 | Katty | so sleepy |
| 15:23.45 | Zeeek | your eyes are getting heavy |
| 15:23.56 | Katty | getting? already there |
| 15:24.03 | Zeeek | your eyes are heavy |
| 15:24.14 | DocAwesome | I'm drinking green tea for that |
| 15:24.24 | Katty | mmmtea |
| 15:24.31 | Zeeek | DocAwesome: you are brilliant, so let me ask you |
| 15:24.36 | DocAwesome | is that anythign like mmmbop? |
| 15:24.41 | Zeeek | and everyone else in the room |
| 15:24.45 | DocAwesome | Zeeek: the answer is most likely something |
| 15:24.53 | Katty | the answer is pie. |
| 15:24.58 | Katty | goes to make tea. |
| 15:25.00 | DocAwesome | the answer is always pie |
| 15:25.03 | DocAwesome | goes to make some pie |
| 15:25.10 | Zeeek | does anyone have a link to a document that states responsibilities for security in a datacebter? |
| 15:25.26 | jaytee | people usually start complimenting me right before they try to borrow money so I'm always on guard when I get a compliment |
| 15:25.29 | DocAwesome | can't say I do... I just put my stuff in Q9 |
| 15:25.40 | ManxPower-work | jaytee: You're awesome! |
| 15:25.45 | DocAwesome | jaytee: yes! |
| 15:27.45 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:31.24 | Katty | has tea |
| 15:32.06 | jaytee | now watching Farscape Season1 Episode 5 |
| 15:32.27 | Katty | oooh i love farscape |
| 15:32.34 | Katty | which episode is it |
| 15:32.41 | jaytee | read back |
| 15:32.52 | Katty | i don't know the numbers |
| 15:32.53 | Katty | how does it start |
| 15:33.03 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:33.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:33.08 | jaytee | Dargo is going through a hyper rage |
| 15:33.26 | jaytee | the title is Thank God It's Friday, Again |
| 15:34.07 | Katty | that's normal for dargo |
| 15:34.12 | Katty | hmmmm |
| 15:34.15 | Katty | googles |
| 15:36.02 | Katty | oh |
| 15:36.03 | Katty | that's episode 6 |
| 15:36.16 | Katty | yes i know that one. |
| 15:36.16 | jaytee | I bought the complete series for $57.99 on Amazon |
| 15:36.28 | Katty | episode 5 is when they find giana |
| 15:37.10 | Katty | 6 is when dargo wants to kill crichton |
| 15:37.23 | Katty | have they gone down to the planet after dargo yet? |
| 15:37.38 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:37.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:37.42 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:37.57 | Katty | rygel has a fun part in this episode :P |
| 15:38.09 | jaytee | you sound like a major fan |
| 15:38.16 | Katty | i am! |
| 15:38.28 | jaytee | one of my friends in Florida named her dog Frell |
| 15:38.30 | Katty | have you seen all the seasons? |
| 15:38.36 | jaytee | nope |
| 15:38.37 | Katty | dog fuck? ^_- |
| 15:38.40 | Katty | interesting |
| 15:38.48 | Katty | ohh |
| 15:38.50 | Katty | i get it |
| 15:39.01 | Katty | that's almost as geeky as me naming my dog riddick |
| 15:39.11 | jaytee | or more geeky |
| 15:39.17 | Katty | yes possibly more |
| 15:39.21 | Katty | ryan says what the frell all th etime |
| 15:39.31 | jaytee | although you've got the whole German-Hobbit thing going |
| 15:39.33 | Katty | as much as i say snickerdoodles |
| 15:39.40 | Katty | yesh. |
| 15:39.52 | Katty | Kaiser Riddick der Kleine Hobbit mit Waggytail |
| 15:40.36 | Katty | ryan said if we got a house in the county that we'd have room for a doberman (= |
| 15:40.47 | jaytee | I actually heard one of the directors at the zoo saying, "What the frak happened to my report?" one day |
| 15:41.13 | Katty | battlestar galactica ya? |
| 15:41.30 | jaytee | yeah, I said, "Big BSG fan?" and he just smiled |
| 15:42.36 | Naikrovek | omg this diesel exhaust is going to kill me |
| 15:43.02 | Naikrovek | diesel air compressor in shop, exhausts to shop, intake air for the heat up in this office, pulls from the shop |
| 15:43.17 | Naikrovek | i.e. death immenent for all office workers |
| 15:43.39 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 15:44.28 | Katty | sounds like someone needs a gas mask for christmas |
| 15:45.00 | Katty | i still need to figure out what i want for christmas. |
| 15:45.19 | Naikrovek | i think the fabrication guys in the shop need to fabricate an exhaust pipe and put it through the wall |
| 15:45.20 | Katty | ryan's helicopter was shipped from hongkong yesterday, apparently |
| 15:45.27 | Naikrovek | ooh helicopter |
| 15:45.37 | Katty | yesh, would you like to see it? |
| 15:45.41 | Naikrovek | please. |
| 15:46.35 | Katty | http://tinyurl.com/ygwwgzh |
| 15:46.52 | Naikrovek | :) |
| 15:46.56 | Naikrovek | want |
| 15:47.01 | Naikrovek | my wife woudl never buy me this |
| 15:47.13 | Naikrovek | i'd sever my spinal cord with it somehow, she thinks |
| 15:47.31 | Katty | boys will always be kids. |
| 15:47.37 | jaytee | I want one 5 times that size that shoots mini Hellfire missles |
| 15:47.42 | Katty | and require kid things to stay at optimum happiness levels |
| 15:47.42 | jaytee | missiles |
| 15:48.01 | Katty | jaytee: do they make those? |
| 15:48.14 | [TK]D-Fender | I want one that's full size carries me as a pilot and shoots FULL SIZE Hellfire missiles |
| 15:48.32 | Katty | good luck with that |
| 15:48.44 | [TK]D-Fender | looks on Craigslist |
| 15:49.25 | Katty | find anything? |
| 15:49.44 | Katty | Naikrovek: i'm thinking about getting him a high quality projector next year |
| 15:49.57 | DocAwesome | yay projectors! |
| 15:49.59 | DocAwesome | big TVs! |
| 15:50.00 | Naikrovek | yes projectors |
| 15:50.05 | Katty | Naikrovek: something bright enough to see with the lights /on/ |
| 15:50.13 | ManxPower-work | T.38 info: http://www.soft-switch.org/t38/ch01.html |
| 15:50.42 | Naikrovek | projector + wiimote + http://hackaday.com/2007/12/09/wiimote-projector-whiteboard/ = awesome |
| 15:50.49 | Katty | looks |
| 15:51.09 | Katty | oooh that looks very fun |
| 15:51.37 | Katty | does it require a wii? |
| 15:52.35 | Zeeek | mais oui |
| 15:52.36 | *** join/#asterisk zoap (n=zoap@53501AB4.static.ziggozakelijk.nl) |
| 15:52.37 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 15:53.15 | zoap | hi there, have a zaptel compile problem. can i fire away? |
| 15:53.28 | Katty | infobot: questions |
| 15:53.29 | infobot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
| 15:53.44 | *** join/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 15:54.07 | DocAwesome | ~ask |
| 15:54.08 | infobot | 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. |
| 15:54.22 | Katty | DocAwesome: that's the one i was looking for |
| 15:54.28 | DocAwesome | :) |
| 15:55.13 | zoap | i try to compile zaptel (zaptel-1.4.12.9.svn.r4653), but get a kernel source is not ok message |
| 15:55.50 | zoap | while the kernel source actually is 2.6.28-11-server kernel, it tells me it is not installed? |
| 15:55.51 | Zeeek | is anyone here? |
| 15:55.54 | DocAwesome | so you have the kernel headers installed for your currently running kernel? |
| 15:56.00 | zoap | yes |
| 15:56.05 | Zeeek | thanks |
| 15:56.06 | DocAwesome | Zeeek: I'm not |
| 15:56.23 | Zeeek | Schroedinger would be proud |
| 15:56.37 | DocAwesome | there is no Schroedinger, and there is. |
| 15:56.44 | Zeeek | Schroedinger's Kat(ty) |
| 15:56.51 | DocAwesome | *cough* |
| 15:57.11 | Zeeek | we're being specific, informative, complete, concise and o |
| 15:57.21 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:57.50 | Katty | waits for eppigy to chime in |
| 15:57.55 | *** part/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 15:58.45 | zoap | never mind, after 42 tries it suddenly does compile... !? must be me then... weird! sorry to have bothered you |
| 15:59.17 | Zeeek | Isn't it obvious? |
| 15:59.21 | Zeeek | 42 |
| 15:59.26 | Katty | hahah |
| 15:59.35 | Zeeek | built in hopness |
| 15:59.40 | jaytee | don't forget your towel! |
| 15:59.42 | *** join/#asterisk ticoit (n=ticoit@201.191.151.147) |
| 15:59.42 | Zeeek | or hipness in English |
| 15:59.50 | Katty | i'd rather hop than hip |
| 15:59.52 | *** join/#asterisk Badrobot- (n=badrobot@76.173.229.89) |
| 16:00.44 | DocAwesome | <-- hip |
| 16:00.54 | Katty | you totally are. |
| 16:01.03 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:01.05 | Naikrovek | wonder when polycom phones will support iax2 |
| 16:01.17 | Katty | as soon as it's profitable |
| 16:01.27 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 16:01.32 | Katty | hi crispers. |
| 16:01.35 | Naikrovek | Katty: no, the wii projector thing does not require a wii. requires a wiimote, a computer, and a projector |
| 16:01.37 | jaytee | <---square.....unless I'm listening to Huey Lewis |
| 16:01.46 | Katty | and an IR device. |
| 16:01.52 | Naikrovek | yes the pen |
| 16:02.01 | Naikrovek | i have all set up at work except for pen |
| 16:02.14 | Naikrovek | gotta find an empty bic pen tube i can use, and a pushbutton and i'll be trying that out |
| 16:02.30 | Katty | sounds exciting. do blog it. |
| 16:02.34 | Katty | ^- in detail. |
| 16:02.37 | [TK]D-Fender | jaytee: Band I may be joining has that on their playlist... |
| 16:02.43 | Naikrovek | it's been blogged to death for the past year or so, not new |
| 16:02.55 | Katty | that doesn't make it any less exciting |
| 16:03.21 | Katty | what would the internet be without redundant information |
| 16:04.05 | jaytee | my buddy just landed a gig as the drummer in a band called "5 minutes to Wapner" |
| 16:04.18 | Naikrovek | lol |
| 16:04.20 | bcrisp | im recording an album |
| 16:04.31 | Katty | jaytee: do you know what kidn of drum setup he uses? |
| 16:04.59 | Katty | bcrisp: i'll broadcast your album |
| 16:05.10 | Katty | bcrisp: if you like |
| 16:05.12 | gaetronik | is there anyway to compile qozap driver on ubuntu 9.10 kernel 2.6.31? |
| 16:05.21 | jaytee | Katty, he's been drumming for over 25 years. he has several sets including an all electric set (which he doesn't like all that much but is more portable) |
| 16:05.41 | DocAwesome | jaytee: I love my electronic kit |
| 16:05.49 | DocAwesome | they are still crazy heavy |
| 16:06.04 | Katty | jaytee: ryan's been talking about getting an electric set...i'd be curious to know what your friend dislikes about it vs a regular set which you have to tune and get heads for, etc |
| 16:06.06 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 16:06.22 | jaytee | I used to work just across the street from the Zildjan headquarters in Norwell, Massachusetts |
| 16:06.47 | DocAwesome | <-- Sabian fan |
| 16:06.47 | jaytee | Katty, I'll ask him about the particulars |
| 16:06.54 | Katty | jaytee: thank you (= |
| 16:07.08 | Katty | DocAwesome: i don't know the first thing about percussion other than they're /loud/ |
| 16:07.10 | Naikrovek | my Guitar hero World Tour drum kit is MIDI and it works as expected. velocity sensitive, even |
| 16:07.11 | bcrisp | Katty: its just instrumental guitar but hey, thanks for the offer |
| 16:07.15 | DocAwesome | I'm looking for a band as well, for anyone who can record and send me stuff to record over top of |
| 16:07.20 | Katty | DocAwesome: which is why i like the idea of ryan getting an electric set with headphone jack |
| 16:07.29 | DocAwesome | Katty: ya -- Rolands are very nice |
| 16:07.33 | DocAwesome | Katty: not cheap though |
| 16:07.36 | Katty | bcrisp: i would be happy to stream it for you regardless. |
| 16:07.38 | bcrisp | cant go wrong with a roland |
| 16:07.41 | DocAwesome | nope |
| 16:07.49 | DocAwesome | I have a Roland TD-7 Turbo brain that needs upgrading |
| 16:07.50 | bcrisp | yamaha.. not bad either |
| 16:07.53 | Katty | DocAwesome: ryan plans on spending around 3 or 4 grand... would that fit a roland set? |
| 16:07.58 | [TK]D-Fender | Katty: Can't grab-stop cymbals, roll off the drum frame to the skin, brushes, etc are iffy... Its playable, but too many drummers I know merely "tolerate" them, but they are very good on space & portability |
| 16:08.01 | DocAwesome | PD-7 and PD-9 pads plus v-drum snare is fine though |
| 16:08.13 | DocAwesome | [TK]D-Fender: I can grab-stop my electronics |
| 16:08.30 | Katty | hmm. |
| 16:08.34 | [TK]D-Fender | DocAwesome: band I jam with used an Alessis DM5 kit.... sounded good and the reviews match |
| 16:08.36 | bcrisp | ive been stuck on playing tommy emmanuel stuff lately... addicting |
| 16:08.43 | Katty | i could see it affecting the technique |
| 16:09.23 | DocAwesome | I have an Alesis DM5 brain -- it's very old |
| 16:09.43 | bcrisp | have you guys used the zoom h2 recorder b4? |
| 16:09.46 | DocAwesome | Katty: just because you play acoustics doesn't mean your technique is any better :) |
| 16:09.53 | bcrisp | its a handy little gadget |
| 16:10.05 | Katty | DocAwesome: wouldn't know. |
| 16:10.12 | Katty | DocAwesome: percussion isn't my area of expertise. |
| 16:10.22 | DocAwesome | I played acoustics for over 5 years before moving to electronics (condos) |
| 16:10.43 | Katty | DocAwesome: what do you figure a roland electronic set would run? |
| 16:10.53 | DocAwesome | Katty: used kit... possibly under $900 |
| 16:10.59 | Katty | oh that's nothing' |
| 16:11.00 | DocAwesome | depends on the brain and pads |
| 16:11.08 | DocAwesome | you can easily spend over $2500 :) |
| 16:11.16 | Katty | do you think ryan could get a fairly decent set for 3 or 4 grand? |
| 16:11.18 | bcrisp | i bought some finger drums from office playground :/ |
| 16:11.35 | Katty | DocAwesome: something respectable |
| 16:11.43 | Katty | DocAwesome: he plays the occasional gig downtown |
| 16:11.46 | DocAwesome | Katty: make sure you get a PD-120 pad far at least he snare |
| 16:11.58 | Katty | takes notes |
| 16:12.03 | DocAwesome | Katty: you could get an amazing kit for that kind of coin |
| 16:12.10 | DocAwesome | Katty: fully decked out V-Drum kit |
| 16:12.11 | Katty | DocAwesome: could you link me some links? |
| 16:12.14 | DocAwesome | Roland V-Drum is the "best" |
| 16:12.19 | Katty | DocAwesome: i'll foward it to ryan |
| 16:12.31 | Katty | DocAwesome: in /query |
| 16:13.10 | DocAwesome | Katty: searchign for something quickly |
| 16:13.14 | Katty | k |
| 16:13.27 | bcrisp | I set up a few agents to play around with the functionality and have noticed that once the agent is connected to a caller in the queue the call ends right at 15 seconds, any clues? |
| 16:13.46 | DocAwesome | Katty: this has more than what he needs: http://drums-percussion.musiciansfriend.com/product/Roland-TD20SX-VPro-Electronic-Drum-Set?sku=582901 |
| 16:14.06 | Katty | looks |
| 16:14.10 | DocAwesome | Katty: this is closer: http://drums-percussion.musiciansfriend.com/product/Roland-TD9SX-Electronic-Drum-Set?sku=491359 |
| 16:14.20 | *** join/#asterisk sasargen (n=chatzill@173-142-206-187.pools.spcsdns.net) |
| 16:14.24 | Katty | hmm |
| 16:14.31 | Katty | so that's the best of the best |
| 16:14.33 | DocAwesome | Katty: then you can just add additional pads |
| 16:14.58 | DocAwesome | Katty: http://www.youtube.com/watch?v=nE2CxdskGII |
| 16:15.08 | bcrisp | do you guys like scat? |
| 16:15.28 | Katty | i'll forward it over to ryan, thanks |
| 16:15.45 | [TK]D-Fender | bcrisp: You set bad timeouts. |
| 16:16.16 | bcrisp | [TD]D-Fender: queues.conf or agents.conf? |
| 16:16.45 | bcrisp | hmm, ill look again while in CLI |
| 16:17.26 | DocAwesome | Katty: another good one: http://www.youtube.com/watch?v=Oct0vs49LMY |
| 16:17.38 | DocAwesome | Katty: those tom pads are the same ones I had -- I have a larger snare pad |
| 16:18.11 | DocAwesome | Katty: this is the one I have for my snare: http://www.roland.com/products/en/PD-120_120RD_120PL/images/PD-120PL_L.jpg |
| 16:18.19 | DocAwesome | great feel |
| 16:18.27 | DocAwesome | anyways, I'll go back to on topic now :) |
| 16:18.42 | Katty | k |
| 16:18.59 | eppigy | 8>~ |
| 16:19.50 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:0:0:0:2) |
| 16:21.18 | *** join/#asterisk psilikon (n=joel@cerberus.vicimarketing.com) |
| 16:22.50 | *** join/#asterisk Pazzo (n=ugelt@p5099e2ec.dip0.t-ipconnect.de) |
| 16:22.53 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:23.21 | psilikon | is there a way to put more than one remote extension behind the same NAT? Say for instance I wanted 10 extensions in 5 remote locations, each location having its own NAT? Without a VPN? |
| 16:23.59 | DocAwesome | psilikon: yes, there is nothing special you have to do other than set nat=yes and create separate configurations in sip.conf for them |
| 16:24.39 | DocAwesome | so you'd just have 10 configurations in sip.conf, and each line would register to asterisk |
| 16:25.30 | Deeewayne | 16 days until Kwanzaa!! |
| 16:25.52 | drfreeze | Anyone know the magic of MWI |
| 16:26.16 | drfreeze | I had a phone that was working and just stopped showing MWI |
| 16:26.24 | p3nguin | reboot it |
| 16:26.34 | drfreeze | tried that |
| 16:26.56 | Katty | wonders why someone would intentionally disable the ide0 channel in bios |
| 16:27.48 | eppigy | for fun and profit |
| 16:28.11 | mcargile | drfreeze is the phone on the same network as the * box or is it remote? |
| 16:28.12 | Katty | good answer, sir |
| 16:28.51 | p3nguin | Maybe they thought they didn't need it. |
| 16:29.24 | *** join/#asterisk sasargen (n=chatzill@173-142-206-187.pools.spcsdns.net) |
| 16:29.45 | bcrisp | [TK]D-Fender: I've checked and don't see anything standing out as far as bad timeouts. I've tested the queue with non-agent and agent members - it appears only to timeout like this when an agent has taken the call (defined in agents.conf0 |
| 16:30.44 | Katty | p3nguin: regardless of whether you need it or not, the port is /there/ |
| 16:30.59 | p3nguin | I just had to fix a system the other day where "the OS won't boot. It gives a kernel panic that it can't find the root file system." I went into the BIOS settings, changed the SATA mode back to 'AHCI' from 'IDE compatibility' the way it was supposed to be... The OS booted on the first shot. |
| 16:31.01 | Katty | p3nguin: why would you go through the trouble of INTENTIONALLY disabling it |
| 16:31.25 | p3nguin | People are weird that way. You know this. :) |
| 16:31.59 | mcargile | Katty: I had to fix that once for someone. They had been told that disabling it would use less resources and make their computer run faster |
| 16:32.14 | psilikon | DocAwesome, would each extension in the sip.conf need to be on a separate port? |
| 16:32.22 | DocAwesome | psilikon: no |
| 16:32.23 | p3nguin | I'm sure you're familiar with the saying about making a better idiot. |
| 16:32.35 | jaytee | my volume license XP w/SP2 CD won't recognize sata drives on Dell Optiplex systems if the mode is set to AHCI |
| 16:32.38 | DocAwesome | psilikon: asterisk listens on port 5060 (by default) and responds on the port the end point sends from |
| 16:32.47 | p3nguin | psilikon: extensions don't go in sip.conf, they go in extensions.conf. |
| 16:32.47 | DocAwesome | psilikon: just try it, you'll see what happens |
| 16:32.49 | bcrisp | jaytee dont get me started on dell machines |
| 16:32.55 | jaytee | fortunately Win 7 doesn't have that issue |
| 16:33.15 | DocAwesome | p3nguin: he's talking about the phones extension number -- which shouldn't be used to identify the devide |
| 16:33.29 | [TK]D-Fender | bcrisp: agents.conf has no role in distributing the calls or giving up, etc |
| 16:33.34 | DocAwesome | psilikon: use something like a MAC address for the identifier, and abstract the person and extension number from the device |
| 16:33.34 | jaytee | bcrisp, oh really? you want me to tell you about the 30 Optiplex SX-280's I've had to replace the mobos on due to a manufacturer defect with bad capacitors? |
| 16:33.38 | [TK]D-Fender | bcrisp: thats all queue call & definition |
| 16:33.39 | p3nguin | docawesome: I should have [tk]d-fender straighten him out. |
| 16:33.48 | DocAwesome | p3nguin: ManxPower-work would be better |
| 16:33.57 | bcrisp | [TK]D-Fender: thats what i thought :: reading :: |
| 16:34.18 | bcrisp | jaytee: i have nightmares... |
| 16:35.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:35.41 | psilikon | DocAwesome, I can get it to work on one remote lan... works great four phones and audio. Yet when I try to do the same thing on another lan I have no audio either way. |
| 16:35.55 | p3nguin | Dell is well-known for having a few models that had bad caps. |
| 16:35.56 | DocAwesome | sounds like a NAT issue |
| 16:36.28 | DocAwesome | psilikon: without additional information there isn't much we can do to debug that issue -- proper configuration of your NAT device and Asterisk should allow it to work fine |
| 16:36.36 | bcrisp | [TK]D-Fender: there's no problem with the caller being dropped from the waiting queue - just once the call is connected with the agent. from there 15 seconds BOOM |
| 16:37.22 | p3nguin | This may be redundant, but did you paste all your configs and a debug? |
| 16:37.32 | *** join/#asterisk Tim_Toady (n=moi@77.49.162.9.dsl.dyn.forthnet.gr) |
| 16:37.34 | psilikon | DocAwesome, currently I have each port of the ATA (sipura spa-2000) configured to use a different port. 5060,5061,5062,5063. Then in the sip.conf I have the corresponding section configured with the same port. |
| 16:37.42 | [TK]D-Fender | p3nguin: Of course not. |
| 16:37.49 | DocAwesome | psilikon: you don't need to do that |
| 16:38.21 | DocAwesome | psilikon: it should all be automatic -- you'll see the ATAs using high numbered ports |
| 16:38.38 | *** join/#asterisk pirulo (n=andres@65.102.99.5) |
| 16:38.40 | *** join/#asterisk Spiegel (n=spike@ip-138-193.interbild.net) |
| 16:38.48 | psilikon | DocAwesome,I saw the ATAs using low ports (1027, 1028, etc) |
| 16:38.48 | DocAwesome | anyways, I have other work related things I need to go work on, so I'm sure other people will be happy to help you |
| 16:39.14 | psilikon | DocAwesome, thanks for your help bro! |
| 16:39.54 | *** join/#asterisk yoshx (n=yoshx@78.114.253.27) |
| 16:40.16 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:45.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:47.43 | *** join/#asterisk dmz (n=dmz@adsl-76-225-155-25.dsl.pltn13.sbcglobal.net) |
| 16:47.58 | Katty | bro? |
| 16:48.47 | Deeewayne | mansier |
| 16:48.54 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 16:50.04 | drfreeze | mcargile: phone is on the same network at the * box |
| 16:54.41 | etfonhomey | Does the Sangoma B600DE fit in a PCI Express x8 slot? |
| 16:55.10 | Katty | call your vendor |
| 16:55.49 | voipmonk | http://www.youtube.com/watch?v=1hrWjkn_DHs&feature=player_embedded - they're here |
| 17:00.49 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 17:02.23 | *** part/#asterisk paulc (n=paulc@64-46-25-2.dyn.novuscom.net) |
| 17:02.26 | *** join/#asterisk paulc (n=paulc@64-46-25-2.dyn.novuscom.net) |
| 17:03.28 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 17:06.30 | bcrisp | hmm |
| 17:09.12 | *** join/#asterisk Tim_Toady (n=moi@77.49.162.9.dsl.dyn.forthnet.gr) |
| 17:10.12 | bcrisp | grr |
| 17:12.21 | Katty | ^- brr |
| 17:12.47 | Katty | i want to go out to lunch today. |
| 17:14.25 | bcrisp | i want a chimi |
| 17:15.50 | Katty | changa? |
| 17:16.04 | Katty | eppigy: let's go have mexican |
| 17:16.06 | bcrisp | ya |
| 17:16.13 | jblack | hi |
| 17:18.32 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 17:19.41 | bcrisp | i think i did something stupid (big gasp of surprise from everyone here) |
| 17:20.12 | carrar | yeah the hand doesn't go in the food disposal |
| 17:20.30 | bcrisp | in my dialplan i have 1, Answer(), n, Queue(queuename), and n, Hangup() |
| 17:20.40 | p3nguin | erroneous spaces? |
| 17:20.50 | bcrisp | when the call is answered by an agent it removes the user from the queue, and i assume it then hangs up |
| 17:21.22 | bcrisp | im gonna comment out the Hangup() and i think it should work |
| 17:21.29 | ManxPower-work | I doubt it. |
| 17:21.31 | p3nguin | no |
| 17:21.34 | bcrisp | no? |
| 17:21.49 | ManxPower-work | The call should stay in the queue until the call ends, then the hangup would be called. |
| 17:21.55 | p3nguin | Furthermore, I would want some additional information to be played before sending a caller into the queue. At least let them know who they have called. |
| 17:22.28 | bcrisp | i was just trying to set up something simple: |
| 17:22.29 | bcrisp | http://pastebin.ca/1709820 |
| 17:22.49 | bcrisp | it works when the queue member is a non-agent, when an agent it hangs up after 30 seconds |
| 17:22.55 | ManxPower-work | bcrisp: that's pretty useless without a copy of the CLI output of a failed call |
| 17:23.02 | bcrisp | it just says user hung up |
| 17:23.29 | bcrisp | ok, ill copy the CLI output |
| 17:26.48 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 17:27.12 | *** join/#asterisk tzafrir (n=tzafrir@bzq-84-111-21-244.red.bezeqint.net) |
| 17:27.42 | bcrisp | ok here is the CLI output: http://pastebin.ca/1709826 |
| 17:28.36 | ManxPower-work | bcrisp: x-lite is hanging up on the caller |
| 17:28.53 | bcrisp | hmm |
| 17:29.14 | ManxPower-work | of course, you did not show the call from the BEGINNING so I can't tell what the cause of this might be. |
| 17:29.29 | bcrisp | oops, one second |
| 17:30.25 | ManxPower-work | also turn the debug on for the CALLING device in addition to the DESTINATION device |
| 17:33.20 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:34.30 | Katty | mmmmmmmmmmmmmexicans |
| 17:34.52 | hardwire | are sexy |
| 17:35.11 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:36.28 | bcrisp | http://pastebin.ca/1709837 should have more detail |
| 17:37.36 | bcrisp | xlite is the caller, agent is on a dif softphone |
| 17:38.17 | bcrisp | oops .. missed your comment on turning debug on for the calling device |
| 17:38.56 | bcrisp | im not sure how to turn on debugging for the calling device :/ |
| 17:40.41 | p3nguin | You place a call to 500@employees, then it goes to 500@incoming, which for some reason (possibly you pressed 2 in an IVR) executes 2@incoming, which then goes to 100@support... where it is finally in a queue. Doesn't this seem a bit messy to you? |
| 17:40.54 | bcrisp | yes |
| 17:40.58 | bcrisp | (learning) |
| 17:41.32 | bcrisp | i was misunderstanding contexts and how to interact between them before.. i think i know how to fix that |
| 17:42.12 | bcrisp | ya 500 gets to a menu (messy gotos), user types 2 so they can be added into a queue |
| 17:43.25 | ManxPower-work | bcrisp: do you hear the one-moment-please sound file? |
| 17:43.41 | jblack | Wow. I'm watching part of that star wars christmas special. It's... mind blowing. http://www.youtube.com/watch?v=VXcb7VPw59s |
| 17:43.59 | bcrisp | ManxPower-work yes |
| 17:44.22 | ManxPower-work | bcrisp: is NAT involved? |
| 17:44.40 | bcrisp | i was under the impression i had to configure xlite clients as NAT = yes |
| 17:44.50 | ManxPower-work | bcrisp: that breaks Asterisk's NAT support. |
| 17:45.12 | bcrisp | ManxPower-work should i set nat=no in the sip.conf and retry? |
| 17:45.17 | ManxPower-work | looks like your signaling address is something like 70.102.242.138 but your audio path is a 10.0.0.whatever. |
| 17:45.31 | ManxPower-work | 10.250.13.108 actuallu |
| 17:45.37 | bcrisp | oh let me see there |
| 17:45.46 | ManxPower-work | bcrisp: turn off NAT support in X-Lite. IS YOUR ASTERISK BEHIND NAT? |
| 17:45.55 | bcrisp | i dont believe so |
| 17:46.03 | ManxPower-work | find out for sure. |
| 17:46.23 | ManxPower-work | because you have to do two different things depending on if the server is behind NAT or not. |
| 17:46.47 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 17:47.13 | bcrisp | besides the messiness did you see anything drastically wrong with the dialplan execution? |
| 17:48.18 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 17:50.09 | eppigy | Katty: okey dokey |
| 17:50.34 | *** join/#asterisk Thor_God (n=Thor@66.237.29.132.ptr.us.xo.net) |
| 17:51.22 | Thor_God | quit |
| 17:52.58 | *** join/#asterisk Thor_God (n=Thor@66.237.29.132.ptr.us.xo.net) |
| 17:54.52 | *** join/#asterisk icyValk77 (n=icyValk7@host81-152-193-216.range81-152.btcentralplus.com) |
| 18:02.03 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 18:05.37 | bcrisp | ManxPower-work: I would guess that if its a network issue, its most likely on my work's configuration |
| 18:06.03 | bcrisp | im going to head home and work from there to reduce the # of variables in this issue |
| 18:06.45 | bcrisp | ManxPower-work: its still interesting to me that if the queue member is a non-agent (SIP/whoever) it works fine, only when its an agent call does it hang up after 30 sec |
| 18:08.00 | voipmonk | odd |
| 18:09.32 | bcrisp | i read elsewhere that 30 second hangups are typical when missing an Answer() somewhere it needs to be |
| 18:10.00 | ManxPower-work | bcrisp: that could easily be a NAT issue, which is why I think you need to resolve the NAT issue before you do anything else |
| 18:10.31 | bcrisp | ManxPower-work: ok, im gonna find out |
| 18:11.06 | ManxPower-work | If the Asterisk server is behind NAT then you need to set localnet= and externip= and need to turn off reinvites. If the client is behind NAT then you need nat=yes in that peer/friend/user sip.conf entry. |
| 18:11.31 | bcrisp | when i toggle nat=no on the xlite sip.conf nothing will come through |
| 18:12.11 | ManxPower-work | If the Asterisk server is behind NAT then you also need to forward the RTP ports. |
| 18:12.17 | ManxPower-work | bcrisp: is the client behind NAT? |
| 18:12.40 | bcrisp | i think it might be .. the office network here is screwy |
| 18:13.06 | *** join/#asterisk tftech (n=Thor@66.237.29.132.ptr.us.xo.net) |
| 18:13.18 | ManxPower-work | bcrisp: you are trying to do brain surgery without knowing knowing anything about the patient. |
| 18:13.35 | bcrisp | ya im leaving the office .. |
| 18:13.45 | bcrisp | im not quite trying brain surgery yet :/ |
| 18:14.48 | voipmonk | ... |
| 18:16.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.41 | bcrisp | server is not behind nat |
| 18:17.48 | bcrisp | client is |
| 18:18.03 | bcrisp | sip.conf has nat=yes for the clietn |
| 18:18.43 | *** join/#asterisk harryv (n=harry@vangberg.name) |
| 18:19.06 | [TK]D-Fender | ~sipnat |
| 18:19.07 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:19.31 | harryv | this is probably totally obvious, but I have no faith in skype, so I'm gonna ask anyways: with skype for asterisk, is it possible to bridge a regular call -> a skype user (that is, not through skypein) |
| 18:19.41 | p3nguin | bcrisp: Where is the server? Where is the client? |
| 18:19.54 | harryv | or can I only do skype-to-skype calls? skype for sip kinda disappointed me in that regard. |
| 18:20.57 | bcrisp | server is 1and1 dedicated server, client is on my office network |
| 18:21.09 | [TK]D-Fender | harryv: Its a channel driver for * |
| 18:21.15 | [TK]D-Fender | harryv: Do the math |
| 18:21.39 | harryv | [TK]D-Fender: I suppose it might have been possible to limit it. |
| 18:22.04 | bcrisp | [TK]D-Fender: thankyou for that link |
| 18:22.05 | harryv | and skype isn't exactly known for letting you call into their walled garden. |
| 18:22.24 | bcrisp | brb |
| 18:22.28 | p3nguin | bcrisp: And the IP address of the server is 70.102.242.138? The IP address of the client is 10.250.13.108? |
| 18:23.55 | p3nguin | I'm not sure how going home is going to change anything. I'm quite certain that you don't have your computer hooked directly to the internet at home. |
| 18:25.36 | p3nguin | Maybe it's just an excuse to go home early. |
| 18:26.48 | *** join/#asterisk sasargen (i=d04a8d70@gateway/web/freenode/x-wnbiymmmqwqnmhae) |
| 18:27.01 | DMeloUK | awk - so close with sip nat, anyone have any exp with multiple phones behind a nat to an asterisk server. one phone calls the other just fine, try to call back and I get congestion message and voicemail - do I need to register the phones on different ports? |
| 18:27.59 | ManxPower-work | DMeloUK: Your NAT router sucks. You might be able to work around this by setting the SOURCE port to different numbers on each SIP phone. The DESTINATION port would stay the same, of course. |
| 18:28.12 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 18:28.30 | ManxPower-work | also remember multiple phones can register to the same account, but only the last registration is used as a destination to send calls to |
| 18:29.06 | bcrisp | p3nguin yes |
| 18:30.03 | DMeloUK | manxpower - they are different extensions just remote to the lan, both can call in and out to the lan and the pstn with no issues, but just one of the phones cannot call the other :) |
| 18:30.05 | bcrisp | p3nguin - (yes to the ip address question) |
| 18:31.00 | bcrisp | my office is point to point vpn to our main office and have strange issues with internet in general .. just thought working from a less complicated network might help |
| 18:31.00 | ManxPower-work | DMeloUK: Ah. Well pastebin the CLI output of a failed call |
| 18:31.20 | p3nguin | bcrisp: Okay, so leave any NAT settings for the server turned off, since it's not NATted; leave any NAT settings ON THE CLIENT disabled, since it breaks Asterisk's configuration; and set nat=yes in the peer/friend/user definition for that phone/user in sip.conf. |
| 18:31.46 | bcrisp | p3nguin ive been looking for NAT config settings on the xlite client but having trouble |
| 18:32.11 | p3nguin | bcrisp: I thought you mentioned earlier that you had turned on some NAT setting on the client. |
| 18:32.34 | p3nguin | bcrisp: Also, for the record, X-Lite is a pile of crap. |
| 18:32.43 | bcrisp | p3nguin: i was misunderstanding -- i have sip.conf with nat=yes for the xlite client |
| 18:33.42 | p3nguin | bcrisp: I've had really good luck with zoiper for Windows, if you're willing to use something other than X-Lite. |
| 18:33.44 | bcrisp | ya sure |
| 18:37.12 | ManxPower-work | bcrisp: The ONLY NAT related thing you might want to enable in X-Lite is NAT Keep-Alive. No other NAT settings should be set. |
| 18:37.15 | bcrisp | installing zoiper |
| 18:37.26 | bcrisp | ManxPower-work: ok ill look at it |
| 18:37.36 | ManxPower-work | p3nguin: All softphones suck |
| 18:39.51 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 18:39.51 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.233.45.dsl.dyn.forthnet.gr) |
| 18:45.50 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 18:46.03 | DMeloUK | manxpower : http://www.pastebin.ca/1709920 |
| 18:47.15 | DMeloUK | manxpower - sip show peers shows ext 201 on port 5060 and ext 202 on port 1024 |
| 18:47.18 | ManxPower-work | DMeloUK: Need another one with the full output from the initial invite to the bye |
| 18:47.28 | ManxPower-work | DMeloUK: that is normal |
| 18:47.46 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:49.25 | DMeloUK | manxpower: http://pastebin.ca/1709928 |
| 18:49.34 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 18:52.33 | ManxPower-work | DMeloUK: your paste is lacking in dialplan output, but it looks like you have a context issue |
| 18:55.20 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 18:55.36 | p3nguin | manxpower-work: Yes, but some suck less than others. |
| 18:55.42 | ManxPower-work | DMeloUK: you realize you are on the wrong channel, right? |
| 18:55.57 | ManxPower-work | #FreePBX is 2 doors down the hall on the left. |
| 18:56.05 | DMeloUK | manxpower - lol |
| 18:56.08 | ManxPower-work | ~freepbx |
| 18:56.08 | 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 |
| 18:56.23 | DMeloUK | I usually just work with asterisk :( |
| 18:56.38 | [TK]D-Fender | And not one of those pastebins shows the device registering, nor the actual DIAL TO IT. |
| 18:56.54 | ManxPower-work | Any yet your pastebins show you are using a FreePBX dialplan. |
| 18:57.26 | [TK]D-Fender | ManxPower-work: And more specifically AsteriskNOW |
| 18:57.53 | ManxPower-work | [TK]D-Fender: All Asterisk GUIs look the same to me. |
| 18:58.13 | [TK]D-Fender | ManxPower-work: H8Shades :) |
| 18:58.29 | ManxPower-work | ?? |
| 18:58.48 | DMeloUK | thanks for the pointers guys - I will look into the context issue and seek help in the appropriate groups |
| 18:58.52 | bcrisp | ugh |
| 18:59.08 | *** join/#asterisk m0e (n=eX@41.196.209.68) |
| 18:59.16 | ManxPower-work | DMeloUK: Specifically I suspect that the extensions.conf context the call is coming into does not have the destination extension in is. |
| 18:59.52 | bcrisp | p3nguin: installed zoiper seems nice.. tried using it with the queue/agent issue but same results |
| 19:00.13 | bcrisp | exactly 30 seconds into the call it hangs up |
| 19:00.28 | p3nguin | bcrisp: I didn't suspect X-Lite was causing the queue problem. |
| 19:00.42 | voipmonk | i'll help you when I get back from the store, bcrisp |
| 19:00.46 | voipmonk | 20 minutes |
| 19:00.46 | p3nguin | bcrisp: manxpower-work, on the other hand, suspected a NAT problem. |
| 19:00.50 | voipmonk | dogs must poop |
| 19:00.52 | bcrisp | p3nguin yes |
| 19:01.21 | bcrisp | ManxPower-work: i ran ifconfig from the server, if i saw internal addresses there it would indicate NAT right? |
| 19:01.47 | m0e | Guys.. I have a question.. I have been banging my head against this issue for a couple days now and I'm just all out of ideas... Now I have an updated asterisknow box with a couple of extentions. To sum up the problem.. Everything seems to work fine except when I use my nokia phone (through 3G) to connect to the asterisk box. Thing is, I have elimiated the problem being hte nokia phone itself since I can connect to gizmo5 using the same phone |
| 19:02.08 | m0e | I'm sure it must be something in the settings that I'm not seeing or that I dont know about |
| 19:02.20 | m0e | Any help is greatly appreciated :) |
| 19:02.21 | p3nguin | An internal address usually means NAT is involved. :) |
| 19:02.27 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:02.35 | m0e | well, yeah.. but I put that machine in a DMZ |
| 19:02.37 | bcrisp | p3nguin, ya i dont see internal ips there |
| 19:02.48 | m0e | and it connects, calls and all... only thing (no voice) |
| 19:03.00 | bcrisp | m0e, what service provider for 3g? |
| 19:03.12 | *** join/#asterisk sasargen (i=d04a8d70@gateway/web/freenode/x-wiourekpiqpdkyyb) |
| 19:03.15 | m0e | vodafone (they arent blocking it) |
| 19:03.19 | bcrisp | ah ok |
| 19:03.22 | bcrisp | :) |
| 19:03.32 | Spiegel | m0e try canreinvite=yes |
| 19:03.51 | Spiegel | this fixed my issued with the voice |
| 19:03.52 | m0e | Really? I read it should be set to no |
| 19:03.55 | m0e | will try that now, thanks :) |
| 19:03.55 | bcrisp | ill be back in about 1/2 hr |
| 19:04.06 | Spiegel | u can try |
| 19:04.12 | Spiegel | np :) |
| 19:06.18 | [TK]D-Fender | It is supposed to be NO. |
| 19:06.25 | [TK]D-Fender | ~sipnat |
| 19:06.26 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:06.28 | [TK]D-Fender | ^^ READ |
| 19:06.33 | Spiegel | there is a good FAQ |
| 19:06.37 | Spiegel | about canreinvite |
| 19:06.58 | m0e | nope.. didnt fix it for me |
| 19:07.07 | m0e | Spiegel, can you point me to that FAQ? |
| 19:07.08 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:07.16 | [TK]D-Fender | m0e: I jsut linked it |
| 19:07.21 | m0e | I'm pretty new at this, so the more I can read the better |
| 19:07.26 | m0e | aaah.. thanks [TK]D-Fender |
| 19:08.35 | *** join/#asterisk yoshx (n=yoshx@78.114.253.27) |
| 19:09.49 | p3nguin | There has to be some cases where canreinvite can be set to yes, otherwise the option would not exist. |
| 19:10.41 | [TK]D-Fender | p3nguin: Indeed. This isn't it |
| 19:11.46 | p3nguin | Does the case never exist within this channel? I've certainly never seen anyone requiring it. |
| 19:12.03 | [TK]D-Fender | p3nguin: very little requirement for it |
| 19:15.51 | *** part/#asterisk harryv (n=harry@vangberg.name) |
| 19:17.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.35 | ManxPower-work | p3nguin: if your client(s) and server are on public IPs or are both on the same lan then you can use reinvites |
| 19:18.41 | p3nguin | hmm |
| 19:20.57 | sasargen | I use SIPAddHeader(Alert-Info: info=alert-autoanswer) to get Aastra 675x phones to autoanswer, but autoanswer always starts the speakerphone. Does anyone know how to force the autoanswer over the headset? |
| 19:21.44 | *** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) |
| 19:23.01 | p3nguin | If, for example, I had a couple of sip users (we'll call them fred and mary) on the same lan and set canreinvite to yes in both of their sip definitiions, they could call each other and reinvite. But what happens if we now introduce another sip user who is outside the lan (we'll call him stan), and we set canreinvite to no in his sip definition? If fred calls stan, will there be a problem? How about if stan calls mary? |
| 19:24.53 | areay | hello... i'm having trouble using the asterisk manager to hangup a call... more specifically i'm having trouble identifying the correct channel to hangup... i've tried using the status command and parsing the output, but it seems like the wrong way of doing things.... is there not an asterisk manager command that can turn, say, a sip extension and the number dialled into the correct channel name? |
| 19:25.11 | ManxPower-work | p3nguin: any call with canreinvite=no set for any of the endpoints should not reinvite |
| 19:25.25 | p3nguin | perfect |
| 19:25.29 | m0e | hmm.. weird.. i got it working, though technically.. i dont know why! I followed that howto (though i had that stuff already in there, but just decided to copy paste then edit) |
| 19:25.36 | ManxPower-work | areay: the thing is there is no such thing as a "sip extension" |
| 19:25.47 | m0e | hmm.. interesting! |
| 19:25.56 | m0e | does the order in sip.conf matter? |
| 19:26.00 | DocAwesome | m0e: it can |
| 19:26.11 | DocAwesome | m0e: type=peer or type=friend? |
| 19:26.11 | *** join/#asterisk wierdo (n=jimmy@77.78.3.197) |
| 19:26.20 | ManxPower-work | You have SIP *devices* and you have dialplan extensions. |
| 19:26.23 | m0e | cause that is the ONLY thing that i notice |
| 19:26.26 | m0e | friend |
| 19:26.32 | DocAwesome | m0e: if matching on an IP address, then yes, order matters if multiple end points have the same IP |
| 19:26.55 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:26.58 | DocAwesome | something is happening that the From header is not being used for matching (or it doesn't actually match anything in sip.conf to match on) |
| 19:27.01 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:27.04 | m0e | cool, thanks a million.. I'm gonna hang around here, try to learn a thing or two.. |
| 19:27.16 | areay | ManxPower-work, ok so if i dial '1901' from 'SIP/1302' how do i hang up the call? |
| 19:27.54 | ManxPower-work | areay: all calls have two channels. The source channel would be SIP/1302-channelnumberinhex |
| 19:28.26 | ManxPower-work | If your extension runs Dial(SIP/1901) then there would be a SIP/1901-channelnumberinhex |
| 19:28.26 | *** join/#asterisk Tim_Toady (n=moi@188.4.76.130.dsl.dyn.forthnet.gr) |
| 19:28.35 | ManxPower-work | Those are really channel instances, not numbers. |
| 19:28.38 | areay | ManxPower-work, how do i get the channel number in hex part? all i've done is originate a call using the manager... surely it would be a lot easier if the originate command gave you the channel number as an output |
| 19:28.54 | ManxPower-work | areay: I have no idea how you list the active channels in manager. |
| 19:29.02 | areay | ManxPower-work, issue a status command |
| 19:29.14 | [TK]D-Fender | areay: AMI throws out NEWCHANNEL events with this info in it.... try paying attention to them |
| 19:29.37 | [TK]D-Fender | areay: And you can poll current channels about a dozen different ways. |
| 19:29.52 | areay | i don't want to have to parse an entire list of current channels |
| 19:30.18 | areay | when i originate a call all i get is "Success, Call successfully originated" or something to that effect |
| 19:30.26 | ManxPower-work | areay: is that worse than processing every NEWCHANNEL event? |
| 19:31.34 | areay | you guys are talking way over my head... let me make it simple: i have a webpage with 2 buttons, dial and hangup... dial works fine, and i don't get any output other than "successful", but hangup doesn't work without the channel name |
| 19:31.40 | areay | <PROTECTED> |
| 19:32.23 | areay | the definition of channel keeps changing too, which is confusing me further |
| 19:32.31 | [TK]D-Fender | areay: So lok it up when you click and jsut do it. No sense in sitting around tracking everything live. |
| 19:32.59 | areay | [TK]D-Fender, you mean parse the entire status output? i suppose it is the easiest way |
| 19:33.13 | areay | it's just frustrating that the Originate command doesn't output the channel numbers |
| 19:33.47 | areay | i dunno... this is a lot of work just for a hangup button |
| 19:34.05 | p3nguin | If you run "core show channels", then the channel is listed in the first column. You can take that channel and then use "soft hangup <that channel>" and it will be hung up. |
| 19:34.25 | [TK]D-Fender | areay: Nearly 2 lines of BASH. Sounds horrible |
| 19:35.09 | areay | [TK]D-Fender, i'm using php and the mxml script that's bundled with the asterisk http server |
| 19:35.26 | areay | and i'm new at ajax so i already have one headache |
| 19:35.31 | areay | but i'll get along with it |
| 19:35.34 | areay | thanks for all your help ppl :) |
| 19:36.31 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:36.35 | areay | oh yeah, does it matter which of the two channels i hangup? |
| 19:36.52 | areay | is it better to hangup the dialling, or dialled channel? |
| 19:36.55 | areay | or both |
| 19:37.19 | [TK]D-Fender | areay: No. |
| 19:37.34 | p3nguin | I would want to hang up the callING channel, but it really shouldn't make a bit of difference. |
| 19:37.47 | areay | i take it the other one is hung up automagically anyway |
| 19:37.48 | p3nguin | dead is dead |
| 19:37.52 | areay | cool |
| 19:38.05 | voipmonk | dead is not dead when its a zombie |
| 19:38.17 | voipmonk | muahahaha |
| 19:38.19 | areay | then it's undead :P |
| 19:38.20 | p3nguin | zombies are not dead, they are undead. |
| 19:39.28 | p3nguin | If a call has already been bridged, is there any way to separate the two legs again? |
| 19:40.16 | voipmonk | to transfer elsewhere and hang one up? |
| 19:40.33 | voipmonk | transfer both ends to separate places? |
| 19:40.51 | voipmonk | I dont think ive ever had to do that outside of having the caller or callee transfer out |
| 19:40.54 | m0e | Sorry, another question.. is externhost buggy? Cause with the same setup, if i comment out externip and add externhost & externrefresh, i'm back to my old problem |
| 19:41.11 | voipmonk | m0e: whats ur problem? :) |
| 19:41.17 | voipmonk | nat? |
| 19:41.40 | m0e | I believe so yes.. If i have externip set, everything works.. with externhost and externrefresh, it doesnt |
| 19:41.43 | [TK]D-Fender | voipmonk: learning to show us. |
| 19:41.48 | *** join/#asterisk CheBuzz_Home (n=CheBuzz_@213-238-66.netrun.cytanet.com.cy) |
| 19:41.49 | m0e | by doesnt i mean (i cant hear the voice) |
| 19:42.30 | p3nguin | voipmonk: I didn't think the question through fully -- I suppose transferring would be one way to separate them. |
| 19:42.38 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 19:42.40 | voipmonk | what proof do you have that rtp is being blocked?, m0e |
| 19:43.07 | m0e | voipmonk, if it was blocked, then wouldnt it NOT work if externip is set? |
| 19:43.16 | p3nguin | m0e: externhost works fine for me. |
| 19:43.31 | voipmonk | do u get audio on both ends when you think its working? |
| 19:43.38 | m0e | yep |
| 19:43.53 | m0e | p3nguin, I'll try again.. maybe I had a typo or something |
| 19:44.14 | voipmonk | so why dont u stick with whats working? |
| 19:44.50 | p3nguin | How can I see if a call between two sip devices did reinvite or not? |
| 19:45.10 | voipmonk | big fat "Re-invite" sip messages |
| 19:45.19 | m0e | well.. I have a dynamic IP (so for testing, its ok to put in the externip clause) but I dont want to write a script to change the externip everytime that my IP changes |
| 19:45.36 | p3nguin | m0e: Use externhost -- that's what it's for. |
| 19:45.56 | voipmonk | m0e: sounds like you found a solution but are too lazy to try it |
| 19:46.01 | m0e | p3nguin, thats the plan :) |
| 19:46.21 | CheBuzz_Home | I have asterisk 1.6 running on OpenWRT. I am trying to send 1800 calls out through gizmo5. But everytime I call, asterisk tells me "No audio format found to offer." These are my peers in sip.conf : http://pastebin.com/d616aaac1 Any reason why I am getting that instead of using ulaw? |
| 19:46.32 | m0e | voipmonk, I usually tend to try the native solutions without me going around it.. That's why I'm trying to get that externhost working :P |
| 19:46.55 | m0e | Thanks alot though everyone.. I'll go have another whack at it :D |
| 19:47.47 | voipmonk | one way to spend your time :) |
| 19:49.31 | *** join/#asterisk tftech (n=Thor@66.237.29.132.ptr.us.xo.net) |
| 19:50.09 | [TK]D-Fender | CheBuzz_Home: You are allowing G.729 and unless you've got licences, * won't transcode and your call will die. Also... you aren't looking at the SIP debug of a failed call <- |
| 19:50.57 | CheBuzz_Home | I have enabled g729 on that peer, but gizmo doesn't allow it. But the both allow ulaw. Shouldn't they agree to do the call in ulaw? |
| 19:52.21 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:2c82:168a:a077:acde) |
| 19:52.41 | [TK]D-Fender | CheBuzz_Home: No, each leg will accept its own preference and * will accomodate them independently. |
| 19:52.43 | [TK]D-Fender | CheBuzz_Home: Now go look at the CALL |
| 19:53.01 | sasargen | areay: see function callme_hangup($exten) at http://www.freepbx.org/trac/browser/freepbx/branches/2.5/amp_conf/htdocs/recordings/includes/callme.php for some PHP code that gets a channel list, does a pattern match to find desired channel, then hangs up that channel. |
| 19:53.02 | voipmonk | hears the whip crash |
| 19:57.32 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 19:58.03 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 20:06.32 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 20:08.44 | x86 | anyone know what this bug is? I think it came from south america but not sure... it's been running around the office all day and someone stomped it... is it poisoinous? ¿cuál es este insecto? ¿cuál es este insecto? |
| 20:09.07 | x86 | http://i.am.leet.org/images/wtf.jpg |
| 20:09.12 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 20:10.23 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
| 20:11.01 | Qwell | x86: whatsthatbug.com |
| 20:11.46 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 20:11.46 | x86 | really? |
| 20:11.48 | x86 | lol |
| 20:11.51 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 20:12.07 | x86 | no way, awesome! |
| 20:12.10 | x86 | thanks man :) |
| 20:12.12 | *** join/#asterisk PabloM (n=user@190.245.131.114) |
| 20:12.30 | PabloM | hi, is anyone using pyst with asterisk 1.6? |
| 20:13.04 | ChannelZ | I have a headache and I'm pyst |
| 20:13.50 | bcrisp | take some midol |
| 20:14.06 | bcrisp | i swear it works better than tylenol |
| 20:14.10 | Katty | i have some, if you need some |
| 20:14.15 | Katty | well of course it does |
| 20:14.19 | *** part/#asterisk CheBuzz_Home (n=CheBuzz_@213-238-66.netrun.cytanet.com.cy) |
| 20:14.20 | Katty | it has tylenol and caffeine |
| 20:14.34 | bcrisp | ya .. ibuprofin does as well right |
| 20:14.37 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 20:14.43 | Katty | no |
| 20:14.57 | Corydon76-dig | You can order discount Midol at 888-344-4860 |
| 20:14.59 | Katty | midol has tylenol, muscle relaxant, and caffeine (stimulant) |
| 20:15.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:15.27 | Corydon76-dig | Free shipping, too |
| 20:15.50 | bcrisp | the caffeine helps |
| 20:16.05 | PabloM | that was really helpfull ¬¬ |
| 20:16.27 | Katty | heh. so i noticed one of our did numbers aren't ringing in right |
| 20:16.40 | p3nguin | hmm |
| 20:17.00 | p3nguin | If one aren't, that's a problem. |
| 20:17.01 | Katty | called the telco |
| 20:17.04 | Katty | they say we don't own it |
| 20:17.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.33 | Katty | i have another number set to roll over to the number we apparently don't own |
| 20:17.45 | Katty | it's all just a big crazy mess |
| 20:18.32 | ChannelZ | why are phone companies so clueless |
| 20:19.08 | Katty | oh i don't think it's the phone company's fault |
| 20:19.15 | *** join/#asterisk canadiancow (n=canadian@CPE001d7edc78be-CM0014e8266d80.cpe.net.cable.rogers.com) |
| 20:19.16 | Katty | probably somewhere at their core they're very organized |
| 20:19.25 | Katty | it's the majority of the people that work for the company that are clueless |
| 20:19.53 | *** part/#asterisk canadiancow (n=canadian@CPE001d7edc78be-CM0014e8266d80.cpe.net.cable.rogers.com) |
| 20:19.57 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 20:19.57 | eppigy | 8[] |
| 20:20.00 | ChannelZ | symantics |
| 20:20.07 | eppigy | sup raden_work |
| 20:20.17 | eppigy | CATHARSIS |
| 20:20.18 | raden_work | notta been working like a mofo |
| 20:20.23 | eppigy | true |
| 20:20.27 | Katty | hi raden |
| 20:20.40 | Katty | i've been setting up my streaming server |
| 20:20.43 | voipmonk | raden - like the god of thunder in mk ? |
| 20:20.47 | ChannelZ | needs lunch |
| 20:20.53 | voipmonk | pronounced Ray den ? |
| 20:21.41 | Katty | in fact, i'll stream my server setup |
| 20:21.45 | Katty | as soon as i have network drivers installed |
| 20:21.51 | raden_work | Katty, what yea streaming |
| 20:21.58 | Katty | nothing at the moment |
| 20:21.59 | eppigy | SHOWER CAM |
| 20:22.02 | raden_work | ha ha |
| 20:22.05 | Katty | no |
| 20:22.12 | eppigy | sorry sometimes I cannot control myself |
| 20:22.20 | raden_work | damn :( |
| 20:22.22 | eppigy | ok all the time |
| 20:22.29 | raden_work | lol |
| 20:22.32 | ChannelZ | it'd be a showering squirrel cam |
| 20:22.54 | Katty | that might happen |
| 20:23.01 | Katty | as soon as i get the camcorder with 200x digital zoom |
| 20:23.04 | ChannelZ | Oooo.. bird bath. There you go. Bow chicka-bow bow! |
| 20:23.12 | x86 | midol++ seriously |
| 20:23.18 | Katty | x86: yesh |
| 20:23.34 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 20:24.22 | Katty | don't tell but i'm using an old rhino server |
| 20:24.53 | Katty | maybe i'l figure out how to change the front lcd display to The Nut House Critter Cam |
| 20:28.23 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 20:29.29 | bcrisp | p3nguin: at home everything is working .. no dropped calls |
| 20:29.44 | *** part/#asterisk PabloM (n=user@190.245.131.114) |
| 20:31.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:37.14 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 20:40.00 | Katty | downloading drivers takes forever |
| 20:40.36 | *** join/#asterisk toddejohnson (n=toddejoh@adsl-70-226-237-31.dsl.spfdil.sbcglobal.net) |
| 20:40.41 | citywok | when some users press 3 in voicemailmain to set their greeting / record their name, all it does is say press * to return to the main menu. why? |
| 20:41.43 | Chainsaw | citywok: Try turning up the core verbosity to 10. |
| 20:41.50 | Chainsaw | citywok: And have a problematic user try again. |
| 20:41.57 | Chainsaw | citywok: It should admit to you what's going wrong then. |
| 20:43.02 | citywok | nothing even at 20 |
| 20:43.08 | citywok | it just prints out the playback prompts it does |
| 20:43.47 | bcrisp | is it printing prompts that you don't hear? |
| 20:43.56 | citywok | i hear them. |
| 20:44.11 | citywok | -- <SIP/1593-00000173> Playing 'vm-starmain.gsm' (language 'en') |
| 20:44.14 | citywok | and i hear it |
| 20:44.37 | bcrisp | i mean, do you see warnings when a problematic user tries it? |
| 20:44.45 | citywok | nothing |
| 20:44.50 | citywok | verbose set to 20 |
| 20:44.59 | Chainsaw | citywok: This amplifier only goes up to 10. |
| 20:45.09 | citywok | it let me record mine, then i tried it again and it wont let me do it now either. |
| 20:45.22 | Chainsaw | citywok: Then it could be something as mundane as file permissions. |
| 20:45.36 | citywok | yea i set chmod -R 777 /var/spool/asterisk/voicemail to check htat |
| 20:46.54 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
| 20:52.07 | tftech | Mine goes to 11 |
| 20:54.51 | DocAwesome | I'll sell you one that goes to 12 |
| 20:55.07 | [TK]D-Fender | docruined the punchline.. |
| 20:55.11 | [TK]D-Fender | DocAwesome: ruined the punchline.. |
| 20:55.15 | [TK]D-Fender | http://xkcd.com/670/ <-------- |
| 20:55.46 | DocAwesome | huh... never saw that |
| 20:55.53 | DocAwesome | I must be just as clever as that guy |
| 20:56.09 | DocAwesome | (that's a lie) |
| 20:57.15 | citywok | [TK]D-Fender: any idea why it wont let users go in to advanced settings? |
| 20:57.45 | [TK]D-Fender | citywok: Maybe because they're listening to a message when they try... |
| 20:57.48 | ManxPower-work | citywok: standby |
| 20:58.14 | ManxPower-work | citywok: IIRC advanced options require the dialout stuff set up in voicemail.conf |
| 20:58.36 | muiro | question. If I have somenone turning on QoS settings on their network, I'm using IAX, can I tell them just to add QoS to 4569 or do I have to give him a range? |
| 20:58.43 | citywok | [TK]D-Fender: it says press 3 for advanced options. press 3 and it says press * to return the main menu immediately |
| 20:59.07 | ManxPower-work | citywok: yup. that's what happens when none of the advanced options are allowed in voicemail.conf |
| 20:59.09 | [TK]D-Fender | citywok: Versions and output please |
| 20:59.28 | citywok | no output with verbose set to 10. SVN 1.6.1 |
| 20:59.32 | ManxPower-work | muiro: IAX2 uses only 4569 |
| 21:00.01 | citywok | ManxPower-work: interesting. i wonder why it worked a couple times randomly. i'll look up these options now and report back. |
| 21:00.17 | ManxPower-work | citywok: enabling them can be a major security issue |
| 21:00.24 | voipmonk | QoS is much like wearing a rented bulletproof vest... you can wear it up intil the check point but from then on you're subject to friendly and enemy fire till u get to the next checkpoint |
| 21:00.46 | bcrisp | i wrote a program to open an AMI session and listen for queue/agent related events. When I have multiple callers in the queue and say the first caller abandons, im not seeing any type of queue status event saying that the other caller's position has changed.. what do i need to look at? |
| 21:00.50 | [TK]D-Fender | voipmonk: there is little more inherently accurate is friendly fire. |
| 21:00.56 | citywok | ManxPower-work: how so? |
| 21:01.09 | ManxPower-work | citywok: the options allow the user to dialout of the system |
| 21:01.28 | Katty | yay nic drivers installed |
| 21:01.40 | muiro | ManxPower-work: many thanks |
| 21:01.44 | ManxPower-work | If the user does not have a secure voicemail password (users never have secure voicemail passwords) then someone could log into the mailbox and make outgoing calls |
| 21:02.20 | citywok | interesting. presently there is no outside voicemail access. nobody ever dials in to it because of visual voicemail on the phone + they are emailed to you. |
| 21:03.44 | citywok | i dont see an option on voip-info that looks relevant to this though |
| 21:04.19 | ManxPower-work | citywok: look at voicemail.conf.sample |
| 21:04.35 | eppigy | LATKES |
| 21:05.20 | citywok | okay, so if i set dialout=somecontext then when i press 3 it says "press 4 to dial out", but i dont get options 1-3 (record greeting, name) |
| 21:05.29 | *** join/#asterisk cmendes0101 (n=none@96.247.7.25) |
| 21:05.38 | ManxPower-work | dialout=fromvm option is one |
| 21:05.41 | *** join/#asterisk ebroad (n=EB@72.11.213.195) |
| 21:05.55 | ManxPower-work | citywok: name, greeting, etc is option 0 not option 3 |
| 21:06.01 | ManxPower-work | "Mailbox Options" |
| 21:06.19 | citywok | wow... epic fail on my part |
| 21:07.07 | citywok | i wonder if it only worked because i fat fingered it. i was reading the tree description of voicemail main from a wiki. wow. sorry for being an idgit |
| 21:07.19 | ManxPower-work | never trust the wiki |
| 21:07.27 | ManxPower-work | trust the config files that came with your version of Asterisk |
| 21:07.44 | bcrisp | ya cuz wiki is a liar |
| 21:09.47 | bcrisp | why does adobe reader have to update every 15 minutes? |
| 21:10.25 | bcrisp | really gets on my nerves |
| 21:10.27 | lanning | because they find security issues that often... |
| 21:10.57 | bcrisp | uninstalls adobe acrobat reader |
| 21:11.36 | ManxPower-work | you can turn off the autoupdates |
| 21:11.40 | p3nguin | bcrisp: Maybe foxit reader would serve you better. |
| 21:11.43 | ManxPower-work | I do that for Foxit Reader |
| 21:12.01 | voipmonk | the new guy at adobe is downloading your cc's |
| 21:12.22 | bcrisp | p3nguin: yah i have both.. just still had adobe installed for some reason |
| 21:16.37 | Katty | infobot: critter cam |
| 21:16.38 | infobot | well, critter cam is http://www.ustream.tv/channel/squirrel-critter-cam |
| 21:16.47 | Katty | ^- |
| 21:16.53 | *** join/#asterisk RonDutt (n=aiontsu@69.7.43.20) |
| 21:17.45 | eppigy | hungry 8[] |
| 21:18.06 | ManxPower-work | eppigy: you eat squirrel? |
| 21:18.22 | Katty | it is not streaming squirrel |
| 21:18.25 | Katty | it's streaming rhino |
| 21:20.00 | beek | Helloooooo, Katty! |
| 21:20.05 | Katty | hi beek |
| 21:20.38 | eppigy | ManxPower-work: i will if it's available |
| 21:21.31 | Katty | hmm. stream is boring |
| 21:21.31 | [TK]D-Fender | steamed squirrel *yum* |
| 21:21.36 | Katty | has a better idea! |
| 21:21.52 | ManxPower-work | eppigy: well go eat all of Kathy's squirrels so she stops going on and on about them |
| 21:22.50 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:24.13 | Katty | :< |
| 21:24.27 | eppigy | that would just be mean |
| 21:24.55 | eppigy | BUT TASTEY |
| 21:25.00 | eppigy | no that is mean |
| 21:26.02 | [TK]D-Fender | Checkout time, later all |
| 21:26.07 | ManxPower-work | eppigy: think of it as "population control" |
| 21:31.20 | eppigy | hunger control |
| 21:34.03 | *** join/#asterisk TommyBotten (n=noosjent@217-14-12-26-dhcp-osl.bbse.no) |
| 21:34.25 | TommyBotten | Has anyone else had problems with asterisk crashing after receiving a T.38 fax? |
| 21:35.17 | Chainsaw | Mine hasn't done that, no. |
| 21:35.28 | Chainsaw | I'm using the SpanDSP-based solution. |
| 21:36.31 | bcrisp | are there any events raised when a queued caller's position changes? |
| 21:37.15 | Katty | anyone know how to reprogram a rhino lcd display |
| 21:40.24 | *** join/#asterisk hakr (i=bryan@element.techlive.tv) |
| 21:45.51 | TommyBotten | Chainsaw: Same here. ast 1.6.1.10 with spandsp 0.0.5. How about you? |
| 21:46.14 | Chainsaw | 1.6.1.11 |
| 21:46.19 | Chainsaw | Confirming SpanDSP version. |
| 21:46.24 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 21:46.51 | Chainsaw | TommyBotten: 0.0.6_pre12 |
| 21:47.12 | TommyBotten | Aha... could I also see your sip.conf related to T.38, pleasE? |
| 21:47.29 | TommyBotten | there are tons of options, and I really don't have any clue ;) |
| 21:47.54 | Chainsaw | You'll have to ask someone else, it's lego star wars time. |
| 21:48.37 | paulc | <3 Lego Star Wars.. although Lego Batman is better I think.. but an 8 year old probably disagrees with me ;-) |
| 21:49.44 | TommyBotten | :( |
| 21:51.28 | ChannelZ | I have lego star wars that I've hardly played. I need to put it in. |
| 21:52.17 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:52.17 | Katty | ah HA! |
| 21:52.27 | Katty | i have found how change the ceros display |
| 21:53.45 | beek | I think that the critter cam has turned into a butt cam. |
| 21:55.12 | ChannelZ | clicks furiously |
| 21:55.35 | Katty | ^_- |
| 21:56.00 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:56.03 | eppigy | lol |
| 21:56.06 | ChannelZ | You have deceived me. There are not butts. |
| 21:56.26 | beek | There was a reflection which made it a butt cam, if only for a few minutes. |
| 21:56.31 | beek | You snooze, you lose! |
| 21:56.45 | ChannelZ | well at least it wasn't squirrel butts |
| 21:56.58 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 21:57.43 | beek | ChannelZ: Would you be dissapointed to know that for a few short moments it was also a cleavage cam? |
| 21:58.18 | ChannelZ | You torture me |
| 21:58.56 | Katty | woah |
| 21:58.58 | Katty | sorry |
| 21:59.00 | Katty | i didn't realize |
| 22:00.35 | eppigy | what did i miss |
| 22:01.11 | ChannelZ | boobs and butts apparently |
| 22:01.33 | bcrisp | you guys should run for public office |
| 22:01.39 | ChannelZ | today is T&A Thursday on #asterisk |
| 22:01.44 | bcrisp | (or join the PGA) |
| 22:02.05 | [TK]D-Fender | bcrisp: I'm probably not the only one who read that as "pubic" at first... |
| 22:02.10 | bcrisp | lol |
| 22:02.18 | bcrisp | wow tiger woods got here fast... |
| 22:03.07 | ChannelZ | I'm surprised Al Sharpton hasn't been out screaming how they're all white women |
| 22:03.10 | bcrisp | anyone like the prospect of using the manager interface who now wouldn't mind bashing it with a shovel? |
| 22:04.00 | bcrisp | thats a little harsh.. its my own ignorance thats bringing frustration |
| 22:05.49 | Katty | what should i put on the middle line? |
| 22:06.43 | Katty | like 'dedicated to dave' |
| 22:08.23 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 22:08.48 | ChannelZ | make it scroll the last line of text in here |
| 22:10.52 | Deeewayne | Katty, you have pest control ads on your critter cam ? |
| 22:11.21 | Katty | it's just google ads |
| 22:11.41 | Corydon76-dig | Topical ads ftw |
| 22:14.04 | *** join/#asterisk cvnet (n=cvnet@69.172.67.161) |
| 22:14.06 | cvnet | hello all |
| 22:15.02 | DocAwesome | Google can tell what is going on based on the video stream -- they are using Google Goggles |
| 22:15.07 | cvnet | exten => 123,n,Background(main-menu,press-0) <-- if i want to play more than one sound file how do i do that? i tried , but it only plays the first found file |
| 22:15.51 | ChannelZ | use & not , |
| 22:16.27 | cvnet | oo ok thanks |
| 22:16.56 | ChannelZ | np |
| 22:17.01 | DocAwesome | core show application Background |
| 22:17.19 | DocAwesome | [Description] |
| 22:17.19 | DocAwesome | <PROTECTED> |
| 22:18.51 | cvnet | thanks |
| 22:19.45 | *** join/#asterisk pirulo (n=andres@65.102.99.5) |
| 22:20.36 | cvnet | WARNING[10286]: file.c:991 ast_streamfile: Unable to open main-menu[ (format 0x4 (ulaw)): No such file or directory |
| 22:20.51 | cvnet | <PROTECTED> |
| 22:20.58 | cvnet | if play main-menu it works fine |
| 22:21.14 | cvnet | i tried main-menu & press-0 it doesnt work |
| 22:21.32 | cvnet | also tried main-menu[&press-0] doesnt work either |
| 22:23.03 | [TK]D-Fender | cvnet: no spaces, no braces |
| 22:23.11 | [TK]D-Fender | cvnet: Stop getting creative |
| 22:23.19 | cvnet | ok thanks |
| 22:23.49 | cvnet | woow it works |
| 22:23.51 | cvnet | i love asterisk |
| 22:23.59 | cvnet | im going to build my very own IVR |
| 22:24.12 | Chris-NB | hi |
| 22:24.15 | voipmonk | you're wasting time talking about it..... |
| 22:24.16 | Chris-NB | anyone trying SRTP with Asterisk? |
| 22:24.19 | voipmonk | do like the shoe says |
| 22:24.22 | voipmonk | Just Do It |
| 22:24.38 | cvnet | ok guys thanks, im going at it now |
| 22:24.40 | ebroad | that would be swoosh |
| 22:24.51 | Chris-NB | or using SRTP? |
| 22:26.16 | ruben23 | hi i have an asterisk server which rubs on 10 callers. and a hosted asterisk service, my connection is 2Mbps for both, are there anyway i can separate the two setup.. |
| 22:26.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:27.14 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 22:29.28 | ecrane | ruben23: can you configure it to rub on me too? |
| 22:30.03 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:31.04 | vlt | Hello. I have to connect an asterisk box to a SIP2PSTN gateway in another room of the building (2 brick walls and 7 meters linear distance) and I can't use ethernet cable. How stable is a wifi connection between two Linksys routers (using WPA2) for VoIP? |
| 22:31.37 | ruben23 | <PROTECTED> |
| 22:31.39 | TommyBotten | vlt: Usually good enough. Just don't let it saturate with other traffic |
| 22:32.37 | russellb | "Usually good enough" -- how comforting :-) |
| 22:33.04 | vlt | TommyBotten: No, it's a dedicated connection. Four simultaneous calls max. |
| 22:33.08 | Qwell | russellb: 9 5s |
| 22:34.04 | TommyBotten | vlt: That will work just fine |
| 22:34.04 | bcrisp | is a caller removed from the queue when they are connected with an agent or when that caller-agen call finishes? |
| 22:34.04 | vlt | TommyBotten: Ok, thank you. |
| 22:34.06 | russellb | Qwell: I don't want to get phone calls, anyway, so 9 5s is pretty good |
| 22:34.12 | Qwell | exactly |
| 22:36.31 | vlt | TommyBotten: How much latency is added usually by such a (WPA2 encrypted) wifi connection? |
| 22:40.13 | Deeewayne | offers vlt a drill with a diamond-tipped bit and an ethernet cable |
| 22:40.31 | ebroad | Deeewayne, i might need to borrow that |
| 22:42.02 | TommyBotten | vlt: That very much depends. Seldom more than 8-10ms |
| 22:42.14 | *** join/#asterisk yahsyn (n=yahsyn@c-24-6-112-203.hsd1.ca.comcast.net) |
| 22:45.50 | vlt | TommyBotten: Thanks. |
| 22:47.21 | *** join/#asterisk korihor (n=korihor@190.205.251.97) |
| 22:47.54 | ebroad | anyone using cdr_syslog and splunk? |
| 22:49.16 | *** join/#asterisk ticoit (n=ticoit@201.191.151.147) |
| 22:51.37 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 22:51.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:53.20 | *** part/#asterisk yahsyn (n=yahsyn@c-24-6-112-203.hsd1.ca.comcast.net) |
| 22:55.03 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 22:56.43 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 23:02.55 | *** join/#asterisk shinao1 (n=shinao1@41.219.221.36) |
| 23:03.47 | ChannelZ | hmmm. |
| 23:04.49 | ChannelZ | needs to find someplace that sells 100W R20 bulbs |
| 23:06.05 | *** join/#asterisk mr_white (n=marco@217.146.140.50) |
| 23:06.16 | mr_white | Hello * |
| 23:06.32 | jaytee | I'd try Lowe's |
| 23:08.16 | jaytee | or Home Depot |
| 23:09.32 | russellb | Lowe's > Home Depot |
| 23:09.32 | cvnet | which file do you edit to create a new voicemail for user? |
| 23:09.46 | russellb | cvnet: a new voicemail account? |
| 23:09.51 | cvnet | yes |
| 23:10.08 | cvnet | users.conf? or voicemail.conf? |
| 23:10.20 | russellb | both. |
| 23:10.26 | russellb | users.conf if you are using users.conf. |
| 23:10.31 | cvnet | hum |
| 23:10.36 | russellb | generally, voicemail.conf. |
| 23:10.48 | cvnet | but how do i know which one i am using? |
| 23:10.54 | cvnet | its the default installation |
| 23:11.02 | russellb | cvnet: have you configured anything? |
| 23:11.42 | cvnet | yes, sip.conf extension.conf and working fine, i have created an IVR now i want to make if the extension dial does not pick up his/her phone to go to his/her voicemail |
| 23:11.46 | ChannelZ | jaytee: neither have any that high unfortunately. |
| 23:11.50 | russellb | then use voicemail.conf |
| 23:13.25 | cvnet | a bit confusing, i installed asterisk gui, but i rather not use it, i created a voicemail for user test it shows htere, but i dont see that in voicemail.conf |
| 23:14.04 | russellb | this is important information! |
| 23:14.10 | russellb | that entire gui is built around users.conf. |
| 23:14.39 | cvnet | a ha |
| 23:14.47 | cvnet | i c that user in users.conf |
| 23:15.15 | cvnet | its better to do it manually in voicemail.conf then using gui right? |
| 23:16.08 | cvnet | @russellb: thanks for everything, I'll do some reading and testing if i get stuck ill be here to ask |
| 23:18.16 | russellb | you're welcome. |
| 23:19.20 | *** join/#asterisk ticoit (n=ticoit@201.191.151.147) |
| 23:20.45 | bcrisp | hmm |
| 23:21.24 | bcrisp | the AMI Queues Action is reporting 0 callers on all queues when there are in fact people waiting |
| 23:22.37 | *** part/#asterisk ecrist (n=ecrist@pdpc/supporter/professional/ecrist) |
| 23:24.42 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 23:27.05 | *** join/#asterisk farrel_antares (n=farrel@187.65.35.252) |
| 23:30.14 | bcrisp | according to al gore, 2km below the service of the earth it is "several million degrees" in temperature |
| 23:30.19 | bcrisp | surface * |
| 23:30.50 | bcrisp | if that were true, wouldn't the earth technically be a star? |
| 23:31.04 | ChannelZ | Could there possible be anyone less credible in the 'climate change' debate than Al Gore? |
| 23:31.12 | bcrisp | i know.. its amazing |
| 23:31.23 | russellb | bcrisp: if that were relevant, wouldn't this be #something-not-asterisk ? |
| 23:31.48 | russellb | that burn was so hot it might be a star |
| 23:32.02 | bcrisp | russellb: sorry |
| 23:32.02 | ChannelZ | We talk about squirrels too |
| 23:32.10 | russellb | lol |
| 23:32.15 | russellb | bcrisp: I'm just messing around, sorry .. |
| 23:32.23 | bcrisp | the steady stream of login / logout messages was putting me to sleep |
| 23:32.29 | russellb | agreed |
| 23:32.39 | russellb | the darkness that comes way too early now was putting me to sleep |
| 23:33.31 | russellb | ~roulette |
| 23:33.32 | infobot | ACTION watches russellb pull the trigger: Click! |
| 23:33.33 | bcrisp | seriously tho.. i can't seem to find the reason why the AMI action: Queues is always reporting 0 callers regardless of the # waiting |
| 23:33.46 | russellb | bcrisp: yes, that does sound wrong. |
| 23:34.35 | bcrisp | launched some sip calls into the queue.. waiting with music, then i telnet'ed in to the management interface and running the Queues command tells me the # of agents correctly but always 0 Callers |
| 23:35.12 | russellb | version? |
| 23:35.13 | thehar | russellb: ! |
| 23:35.19 | russellb | thehar: OMGOMGOMGOMG HI! |
| 23:35.23 | thehar | ZOMFG |
| 23:35.27 | thehar | HAI |
| 23:36.19 | bcrisp | 1.6.1.11 |
| 23:36.35 | ChannelZ | bcrisp: hmmm mine shows the queues right |
| 23:36.48 | russellb | bcrisp: well ... we do have a 1.6.1.12-rc1 out ... |
| 23:36.59 | russellb | you could try that, but that bug does not sound familiar |
| 23:37.03 | bcrisp | i searched the bugs listing but didnt see any |
| 23:37.32 | ChannelZ | bcrisp: what does yours show? |
| 23:37.38 | bcrisp | let me pastebin |
| 23:37.38 | bcrisp | one sec |
| 23:37.42 | russellb | bcrisp: does "queues show" at the CLI produce the same result? |
| 23:37.54 | bcrisp | russellb: ill try that now |
| 23:38.16 | ChannelZ | I just did a test although the box I did it on is 1.4 |
| 23:38.24 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 23:39.03 | bcrisp | russellb: queue show works properly |
| 23:39.07 | bcrisp | showing 2 callers |
| 23:39.15 | russellb | innnnnnnnnnteresting |
| 23:39.27 | bcrisp | ill re-run the manager action now too |
| 23:40.29 | bcrisp | yikes now it is working |
| 23:40.34 | russellb | ! |
| 23:40.38 | russellb | a heisenbug! |
| 23:40.43 | bcrisp | that is annoying |
| 23:41.08 | bcrisp | heisenbug lol |
| 23:41.11 | russellb | that syntax for showing callers is horrendous |
| 23:41.23 | russellb | is reading code ... |
| 23:41.47 | russellb | it's totally not valid AMI syntax, but oh well |
| 23:41.49 | ChannelZ | the output you mean? |
| 23:41.52 | russellb | yes |
| 23:42.03 | russellb | it's a caller on each line, right? |
| 23:42.11 | russellb | and not in a proper key: value pair... |
| 23:42.19 | ChannelZ | yeah and a bunch of other formatting |
| 23:42.25 | russellb | shudders |
| 23:42.37 | russellb | I apologize for that, heh. |
| 23:42.54 | bcrisp | here's my output: http://pastebin.ca/1710308 |
| 23:43.34 | ChannelZ | It's quite human-friendly |
| 23:43.40 | bcrisp | lol |
| 23:43.47 | bcrisp | my parser is screaming |
| 23:44.33 | Deeewayne | ~roulette |
| 23:44.34 | infobot | ACTION watches deeewayne pull the trigger: Click! |
| 23:44.35 | bcrisp | i actually wouldnt mind a checksum mechanism be in place |
| 23:44.43 | Deeewayne | woot |
| 23:44.51 | bcrisp | ~roulette |
| 23:44.52 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 23:45.02 | bcrisp | ~woot |
| 23:45.03 | infobot | \o/ |
| 23:45.08 | Deeewayne | offers bcrisp the bottle of tequila |
| 23:45.14 | Deeewayne | s/tequila/vodka |
| 23:46.07 | farrel_antares | Hello guys, I've just installed asterisk together with a local telephony operator to be able to do calls to PSTN network. To my clients I put an grandstream ATA GXW4004 or GXW4008. The service's packet are as following: 2 phones and 1 link internet. The clients get one of these lines, for example the line2 to use with credit cards machines. The problem is: I have a terrible problem with VISA, I got some errors like: "No line", "Conection Error", |
| 23:46.07 | farrel_antares | and so on. But with the others credit cards I have no problem like this, everything works very well. Does anyone already here had a problem like this? |
| 23:47.25 | bcrisp | implodes |
| 23:47.44 | Deeewayne | I also have terrible problems with VISA |
| 23:48.20 | farrel_antares | Deeewayne, I'm having serious problems, because I have about 100 clients. |
| 23:49.18 | bcrisp | im having serious problems because i owe them a LOT of money |
| 23:49.50 | *** join/#asterisk ruied (n=ruied@bl7-211-221.dsl.telepac.pt) |
| 23:50.26 | farrel_antares | Deeewayne, I've got to reduce the problems with some settings on ATA, such as: Jitter, Gain, and changing some credit cards machines |
| 23:50.50 | Deeewayne | ~roulette |
| 23:50.51 | infobot | ACTION watches deeewayne pull the trigger: Click! |
| 23:52.05 | Deeewayne | farrel_antares, the same line2 works when you call someone other than VISA ? |
| 23:52.34 | farrel_antares | Deeewayne, yes. perfectly. |
| 23:53.01 | farrel_antares | Deeewayne, the others credit cards works very well. |
| 23:53.28 | Deeewayne | and its voip from * to the ATA ? |
| 23:57.06 | farrel_antares | Deeewayne, I don't understood your ask. |
| 23:57.22 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:59.55 | farrel_antares | Deeewayne, I'm together with a telephony operator that gave me a SIP trunk. |
