| 00:05.52 | *** part/#asterisk pietro1 (~pietro@88-149-226-183.dynamic.ngi.it) |
| 00:07.33 | *** join/#asterisk zurdo13 (~Jose@201.130.0.10) |
| 00:07.57 | tompaw | Evening guys. |
| 00:08.17 | tompaw | Anyone using web sip phones? Or maybe Flashphoner in particular? |
| 00:09.39 | SeRi | waz up p3nguin! |
| 00:09.46 | SeRi | waz up [TK]D-Fender! :) |
| 00:16.46 | *** join/#asterisk pushpop (~pushpop@pool-173-77-211-176.nycmny.fios.verizon.net) |
| 00:17.39 | pushpop | I was wondering how would I make asterisk register to flowroute for incoming and outgoing calls and keep all my phones on the lan still register to the pbx on port 5060? |
| 00:20.45 | [TK]D-Fender | pushpop: Same answer as we've given you for 3 days now. |
| 00:20.46 | ChannelZ | ..just do it |
| 00:21.25 | pushpop | sip proxy |
| 00:21.28 | pushpop | is what ur saying |
| 00:21.33 | [TK]D-Fender | Siad |
| 00:21.35 | [TK]D-Fender | said |
| 00:21.38 | [TK]D-Fender | repetitively |
| 00:21.53 | ChannelZ | No.. make your phones reg to *, reg * to flowroute. Edit dialplan as desired. |
| 00:22.08 | [TK]D-Fender | ChannelZ: No.... he has port issues |
| 00:22.08 | ChannelZ | It's sort of how it usually works. |
| 00:22.28 | pushpop | calls are dropping on outbound calls |
| 00:22.31 | ChannelZ | Blocked? |
| 00:22.34 | [TK]D-Fender | yes |
| 00:22.40 | pushpop | flowroute recommends to use their alt port 5160 |
| 00:23.01 | ChannelZ | or find a less shitty ISP if possible |
| 00:23.14 | pushpop | haha yea that would be the answer |
| 00:23.21 | pushpop | but I'm stuck atm with this isp |
| 00:23.41 | [TK]D-Fender | pushpop: Same answer as always. * won't listen on 2 ports. |
| 00:24.08 | ChannelZ | Why is it a problem to make the phones reg on a different port? |
| 00:24.11 | [TK]D-Fender | Not going to happen. Shove another * in between. Run a SIP proxy that can listen on end and parse out the relay to the other. Whatever, |
| 00:24.23 | ChannelZ | Are they massively unconfigurable? |
| 00:24.40 | pushpop | [TK]D-Fender i appreciate your time but I dont know much about sip proxy nor how to configure |
| 00:24.43 | *** part/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 00:26.12 | [TK]D-Fender | pushpop: And that fact doesn't change your list of options. |
| 00:26.26 | [TK]D-Fender | pushpop: Run some other instance of something else to sit in between. |
| 00:26.33 | pushpop | do you know of a how-to |
| 00:26.41 | pushpop | that will explaint to me how to do this |
| 00:26.48 | tompaw | I need an idea. I am inside an extension and suddenly applicationmap triggers an action - for example a macro call. How do I keep the value of ${EXTEN} through this action? |
| 00:26.57 | [TK]D-Fender | Specifically, no. |
| 00:27.01 | ChannelZ | Can you send out to the world on 5060, but not receive? Or neither? |
| 00:27.14 | [TK]D-Fender | shoving another * in between is certainly easy enough |
| 00:27.28 | pushpop | so say I have a remote asterisk box |
| 00:27.33 | pushpop | on a different public ip |
| 00:27.38 | [TK]D-Fender | ChannelZ: Outside = 5160 (both ways), inside = 5060 only both ways |
| 00:27.51 | [TK]D-Fender | pushpop: I never said on another IP |
| 00:27.55 | [TK]D-Fender | (public) |
| 00:27.58 | [TK]D-Fender | 2 boxes onsite |
| 00:28.09 | pushpop | ugh |
| 00:28.23 | ChannelZ | Just wondering if it could be hotwired with port forwarding |
| 00:28.45 | [TK]D-Fender | ChannelZ: Maybe... not sure on it, but maybe... |
| 00:28.48 | ChannelZ | I did something like it to work around Comcast's blocked samba once a long time ago |
| 00:29.20 | SeRi | Are we at this again? |
| 00:29.31 | *** join/#asterisk kaushal (~kaushal@14.99.156.187) |
| 00:29.46 | kaushal | Greenlight: Hi |
| 00:30.17 | p3nguin | Why would you need to mess with asterisk to use an ITSP's alternate port? |
| 00:30.47 | pushpop | p3nguin? |
| 00:31.20 | p3nguin | (1830.47) <pushpop> p3nguin? <---- I don't understand your question. |
| 00:31.29 | SeRi | looks like he is trying to register multiple devices with one public IP |
| 00:31.41 | p3nguin | That's not going to happen. |
| 00:31.51 | [TK]D-Fender | No |
| 00:31.54 | SeRi | I am guessing here |
| 00:31.57 | [TK]D-Fender | that is not what he's doing |
| 00:31.59 | SeRi | or blocked port |
| 00:32.04 | [TK]D-Fender | Guessing.... wrongly |
| 00:32.13 | pushpop | outbound calls are being dropped |
| 00:32.22 | pushpop | something with ISP and port 5060 |
| 00:32.34 | SeRi | who is your isp? comcast? |
| 00:32.38 | [TK]D-Fender | "something"? |
| 00:32.41 | p3nguin | So use the ITSP's alternat port. |
| 00:32.47 | pushpop | Cablevision Optimum Online |
| 00:32.49 | [TK]D-Fender | How is a port intermittent? |
| 00:33.48 | SeRi | pushpop, its illegal for isp's to block ports that would cause a service interruptions such as voip.... or something along those liens I once heard. |
| 00:34.08 | pushpop | what scares me |
| 00:34.11 | pushpop | is the modem |
| 00:34.20 | SeRi | model? |
| 00:34.24 | pushpop | has fxo ports for the voip service |
| 00:34.32 | pushpop | their* |
| 00:34.35 | p3nguin | While it is a denial of service to block ports, the customer agreements often have you contract yourself out of any recourse. |
| 00:34.37 | pushpop | so maybe the modem is causing issues |
| 00:34.38 | SeRi | pushpop, you had the same issue I had with comcast |
| 00:34.57 | SeRi | 5060 was been mangled by comcast. |
| 00:35.06 | SeRi | Because I use there phone service |
| 00:35.12 | p3nguin | So their modem is intended to connect to two telco lines. Interesting. |
| 00:35.13 | pushpop | and you had issues with dropped calls and one way audio issues? |
| 00:35.26 | ChannelZ | There's no options in the modem to turn that crap off? |
| 00:35.33 | pushpop | no =/ |
| 00:35.38 | SeRi | They fixed it after a wireshark |
| 00:35.59 | SeRi | pushpop, no. I just couldnt use that port what so ever |
| 00:36.10 | pushpop | i can use it |
| 00:36.15 | pushpop | just having intermittent issues |
| 00:36.16 | p3nguin | Now why would a cable modem have ports to connect to telco lines? |
| 00:36.20 | SeRi | pushpop, audio does not travel over 5060 |
| 00:36.28 | pushpop | yea i know trp |
| 00:36.31 | pushpop | rtp |
| 00:36.36 | SeRi | so whay are you asscoiating the two |
| 00:36.51 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 00:36.52 | SeRi | why* |
| 00:36.54 | pushpop | i guess i shouldnt |
| 00:37.00 | p3nguin | Have we seen the evidence yet? |
| 00:37.06 | pushpop | but alot of outbound calls dont go through |
| 00:37.07 | ChannelZ | If you are able to make and receive calls, one-way audio or not, it's not 5060 that is your problem |
| 00:37.15 | pushpop | got cha |
| 00:37.16 | SeRi | ^^ |
| 00:37.35 | SeRi | exactly what ChannelZ said. and as p3nguin said where is yuor evidence? |
| 00:37.51 | SeRi | sip debug? |
| 00:37.54 | pushpop | I guess I have no hard evidence at the moment |
| 00:38.02 | pushpop | i have logs |
| 00:38.04 | SeRi | Yuo using asterisk? |
| 00:38.08 | pushpop | haha |
| 00:38.09 | pushpop | ya |
| 00:38.09 | SeRi | you* |
| 00:38.12 | kaushal | can somebody please suggest me about http://pastebin.ubuntu.com/739804/ |
| 00:38.21 | p3nguin | Okay, good luck with this. I have better things to do. |
| 00:38.23 | SeRi | Than: sip set debug on |
| 00:38.27 | kaushal | i dont see anything in /tmp/outputfile |
| 00:39.07 | SeRi | p3nguin, +1 and looking back at the logs he mislead [TK]D-Fender and the others I think... |
| 00:39.17 | SeRi | ~faith |
| 00:39.17 | infobot | Telephony is not faith-based. Look. Always look, and then show. |
| 00:39.30 | pushpop | sorry guys |
| 00:40.00 | kaushal | Greenlight: please suggest me about http://pastebin.ubuntu.com/739804/ |
| 00:40.02 | SeRi | pushpop, just gather some sip debug and use pb and the vets here can help you |
| 00:40.19 | ChannelZ | kaushal: I don't think you can just do that, with the >> in a system call |
| 00:40.20 | SeRi | people here are usually very helpful. |
| 00:40.27 | pushpop | OK will do |
| 00:41.02 | kaushal | ChannelZ: any way out ? |
| 00:42.26 | ChannelZ | I guess you could make a shell script and call that instead, let the shell script do the redirection |
| 00:42.49 | ChannelZ | although wait.. that file is appearing when you run past that part of the dialplan? |
| 00:42.58 | ChannelZ | it's just empty? |
| 00:44.41 | kaushal | ChannelZ: yes |
| 00:45.06 | kaushal | ChannelZ: does that file shows up in that pastebin dialplan reload ? |
| 00:45.07 | ChannelZ | huh.. well that's actually more than I expected to happen so hmmm... |
| 00:45.24 | ChannelZ | eh? |
| 00:45.52 | SeRi | p3nguin, I decided what I am goign to do as far as a new asterisk server. |
| 00:46.46 | ChannelZ | Intersting, I just tried System(echo barf > /tmp/outfile) and it worked here. Shows you what I know |
| 00:47.11 | ChannelZ | My guess is that your variable expansions aren't working there. Testing.. |
| 00:47.13 | SeRi | I am going to build it my self. Instead of buying a fully built system. Looking at some nice cases online right now, |
| 00:47.41 | ChannelZ | nope that worked too |
| 00:48.52 | [TK]D-Fender | kaushal: How is it you show us that PB ... tell us you don't see anything in the file (after making us guess what the problem is) .... and then don't even have a call in there to show us the problem? |
| 00:49.33 | pushpop | Could SIP ALG being enabled on the router be causing issues with asterisk as far as outbound calls dropping? |
| 00:49.36 | ChannelZ | Yah that's my only guess.. did you actually make a call that hit that extension? The syntax is working here. |
| 00:49.52 | kaushal | ChannelZ: yes |
| 00:49.58 | [TK]D-Fender | pushpop: Should always be disabled. And whoe know. Go look at a call fail. |
| 00:50.10 | [TK]D-Fender | kaushal: Where is the failure? |
| 00:50.14 | pushpop | well its enabled |
| 00:50.18 | pushpop | ill disable |
| 00:50.33 | kaushal | I dont have any failures |
| 00:50.41 | ChannelZ | oh, well duh... your dialplan isn't going to continue after the DIal when you hangup |
| 00:50.42 | SeRi | pushpop, now make a test call to replicate the issue |
| 00:50.47 | [TK]D-Fender | kaushal: Where is the call? I don't see anything happening at all |
| 00:50.53 | kaushal | the concern is i dont see anything in the outputfile |
| 00:51.01 | SeRi | and get the logs and paste thim in pastebin |
| 00:51.15 | kaushal | ok |
| 00:51.20 | [TK]D-Fender | kaushal: You are showing me a gun and expecting the sigth of it to tell me why my neighbour decided to kill his wife with it |
| 00:51.25 | [TK]D-Fender | sight* |
| 00:51.45 | ChannelZ | kaushal: Add ,,Fg do your Dial statement |
| 00:52.10 | [TK]D-Fender | ChannelZ: SHUSH YOU. I want my evidence! :p |
| 00:53.03 | ChannelZ | OK Columbo :) |
| 00:53.34 | kaushal | [TK]D-Fender: http://pastebin.ubuntu.com/739821/ |
| 00:54.17 | ChannelZ | that doesn't show anything |
| 00:54.18 | [TK]D-Fender | kaushal: Nowhere in that twisted originate to dyou ever hit the dialplan |
| 00:54.33 | [TK]D-Fender | kaushal: What does that have to do with extensions.conf? It doesn't even use it |
| 00:54.35 | ChannelZ | Even the number you dialed doesn't match, even if you didn't use originate :) |
| 00:54.47 | [TK]D-Fender | host0039*CLI> channel originate DAHDI/g2/9930963298 Application MP3Player /home/kaushal/obd-demo.mp3 <---- nothing about dialplan here |
| 00:54.58 | kaushal | ok |
| 00:55.27 | kaushal | Apologies |
| 00:55.44 | kaushal | not sure i understand that |
| 00:55.56 | [TK]D-Fender | kaushal: that originate doesn't use the dialplan. |
| 00:55.58 | kaushal | Any way to test it ? |
| 00:56.31 | ChannelZ | kaushal: you're connecting a channel dierctly to an application via channel originate, so nothing is happening in your dialplan. |
| 00:56.40 | [TK]D-Fender | kaushal: Is this hard to understand? There are no "step numbers" for it to process. It dials a device. not by matching a number dialed into the dialplan. It jsut goes OUT. And when it answers it executes that 1 dialplan app DIRECTLY. |
| 00:56.51 | [TK]D-Fender | kaushal: No where is it executing a single line from extensions.conf |
| 00:57.33 | kaushal | understood now |
| 00:57.37 | ChannelZ | kaushal: Do you have a softphone or a hardphone setup? |
| 00:57.58 | kaushal | Its just configured as Outbound dialling |
| 00:58.20 | kaushal | There is no softphone or hardphone setup |
| 00:58.32 | [TK]D-Fender | Go change that |
| 00:58.33 | ChannelZ | Spam calls |
| 00:58.37 | *** join/#asterisk Beetlejooz (~IceChat77@12.237.61.93) |
| 00:58.49 | kaushal | basically i am using AMI using java |
| 00:58.57 | Beetlejooz | Does anyone know of a good aggregate SIP provider? |
| 00:59.11 | kaushal | apologies again |
| 00:59.26 | kaushal | when does dialplan come into action |
| 00:59.41 | kaushal | please help me understand |
| 00:59.50 | ChannelZ | when you have actual calls flowing through the system |
| 01:00.14 | ChannelZ | You're basically using Asterisk as an audio player over the phone it looks like |
| 01:00.30 | [TK]D-Fender | Which is fine |
| 01:00.38 | ChannelZ | Sure |
| 01:00.39 | [TK]D-Fender | If you were doing a valid test |
| 01:01.01 | ChannelZ | But if you're doing this through AMI in the first place, I think you can get all the information you need via AMI as well to do your logging. |
| 01:01.23 | ChannelZ | In fact, asterisk CDR is already doing what your example was trying to log.. |
| 01:01.37 | kaushal | ok |
| 01:01.40 | Beetlejooz | Anyone? :) |
| 01:01.55 | [TK]D-Fender | CDR doesn't normally log unanswered calls. Nor does it include hangup codes, etc |
| 01:02.38 | ChannelZ | hmm |
| 01:02.55 | kaushal | [TK]D-Fender: so in my case how do i log all details for http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 01:03.12 | kaushal | unanswered etc...... |
| 01:03.37 | ChannelZ | ~itsplist-us |
| 01:03.37 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 01:06.25 | Beetlejooz | I need one that aggregates SIP services. |
| 01:06.52 | [TK]D-Fender | Beetlejooz: Providers aren't there to collect other provideres services for you |
| 01:07.25 | [TK]D-Fender | Beetlejooz: You actual needs (imagined or otherwise) are very unclear. |
| 01:08.21 | kaushal | [TK]D-Fender: where can i find details about unanswered calls ? |
| 01:11.15 | SeRi | p3nguin, you avail? |
| 01:12.07 | Beetlejooz | TK: I was told that the most reliable way to deploy SIP is through an aggregate provider. |
| 01:13.32 | Beetlejooz | A provider who resells mutiple SIP carrier services and can offer failover when a provider goes down |
| 01:14.12 | Beetlejooz | Preferably a carrier in a tier 2/3 position |
| 01:14.14 | [TK]D-Fender | And when your agregate goes down, everything goes down. Single point of failure.... |
| 01:14.34 | SeRi | Beetlejooz, [TK]D-Fender beat me to it. |
| 01:14.36 | Beetlejooz | A true aggregate has multiple locations |
| 01:14.40 | [TK]D-Fender | Do your own fialover <- |
| 01:15.46 | Beetlejooz | I can't imagine someone selling aggregate services all from the same building. That would be...silly. |
| 01:16.31 | Beetlejooz | Joe Backhoe could ruin everyone's day. |
| 01:16.41 | [TK]D-Fender | Any major ITSP would have multiple backends already as it is. |
| 01:17.02 | p3nguin | *yawn* |
| 01:17.26 | Beetlejooz | I'm 20M up/down through vitelity and I still get congestion. |
| 01:17.37 | [TK]D-Fender | It's the emergency backup redundancy department of redundancy! |
| 01:18.19 | SeRi | p3nguin, bad day today? |
| 01:18.21 | SeRi | :) |
| 01:21.59 | SeRi | dont you hate when family do things without asking help and when they tell you is to late to fix there fuck up. |
| 01:22.13 | SeRi | argh!!!!! |
| 01:24.18 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:24.55 | p3nguin | Fuck 'em. Tell them to learn from their mistake and ask you next time. |
| 01:25.08 | SeRi | YES! |
| 01:25.39 | SeRi | exactly! I want to scream to the little bastard! |
| 01:25.49 | p3nguin | Let 'er rip. |
| 01:26.38 | SeRi | opens up a fucking can of ass whooping! |
| 01:27.38 | SeRi | so true though.... I am not going to stress.... let them deal with it... |
| 01:27.44 | LiuYan | ~odbc |
| 01:27.44 | infobot | [odbc] Open Data Base Connectivity |
| 01:27.46 | SeRi | let them learn. |
| 01:29.47 | p3nguin | Be sure to remind them about the mistake at every opportunity. |
| 01:30.48 | [TK]D-Fender | BBL |
| 01:31.27 | SeRi | cya [TK]D-Fender |
| 01:31.44 | SeRi | p3nguin, I sure will. |
| 01:32.34 | SeRi | every time from now on. and I know it will happen again soon. and ill be like "fu!" didnt need my help before dont need it now. |
| 02:15.42 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:21.05 | SeRi | p3nguin, can IAX2 detect faxes? |
| 02:21.20 | p3nguin | Not to my knowledge. |
| 02:21.50 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 02:21.55 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 02:22.05 | SeRi | I see. is it worth it using IAX2 over SIp when I there is not much bandwidth consumption? |
| 02:22.54 | p3nguin | I can think of only two main reason to use IAX2: NAT traversal, and trunking to save bandwidth. |
| 02:23.19 | SeRi | I see. |
| 02:23.31 | p3nguin | But since you are not doing any fax detection anyway, I don't understand the intial question. |
| 02:24.38 | SeRi | True. I guess I was just curious. because when I started with asterisk I was using IAX2 and I think I switch away from it because of faxing. some where I read IAX2 can not recive faxes. |
| 02:24.59 | p3nguin | I bet you added faxdetect back to your conf after I removed it, too. |
| 02:25.11 | SeRi | No sr. |
| 02:25.17 | SeRi | Its all is it was. |
| 02:25.35 | p3nguin | I use IAX2 for voice, and SIP/RTP for fax. |
| 02:26.08 | SeRi | I see. |
| 02:26.14 | SeRi | Makes sense. |
| 02:26.27 | p3nguin | Well, on VoIP.ms I use IAX2 for voice... I use SIP for voice on some other providers, which lack IAX2 support. |
| 02:26.50 | SeRi | I might do something similar. I need to wait for my month with CC to be over and decide what Ia m going to do for outgoing calls. |
| 02:27.09 | SeRi | I see. yea CC does not support IAX |
| 02:27.15 | SeRi | IAX3* |
| 02:27.18 | SeRi | lol |
| 02:27.21 | SeRi | IAX2* |
| 02:27.36 | p3nguin | IAX3 - IAX = IAX2 |
| 02:27.44 | SeRi | :) |
| 02:28.40 | SeRi | A rep from voip.ms told me that they are working on adding fax support on there services as well as web enable voicemail. |
| 02:29.09 | SeRi | nether which I care for. but thought I pass it. |
| 02:30.03 | SeRi | My only need for SIP would be faxing. as soon as the month is done ill compare the two bills and see where I am ahead and just switch for good. |
| 02:31.04 | SeRi | Though I did score a did from CC for $2.28 a month with unlimited outbound calling |
| 02:31.20 | SeRi | I mean incoming* |
| 02:31.51 | SeRi | with some offer they had regarding unused DID'a |
| 02:31.57 | SeRi | 's* |
| 02:32.49 | SeRi | http://www.callcentric.com/dids/phone_number_sale |
| 02:45.42 | SeRi | in any case I dont need CC for DID |
| 02:47.12 | SeRi | ~magicjack |
| 02:47.34 | SeRi | The bot new better lol |
| 02:47.46 | SeRi | I guess thats how bad they suck. lol |
| 02:53.11 | SeRi | I been thinking of canceling sat and moving to something like google tv... :/ |
| 02:53.42 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 02:54.45 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 02:55.29 | p3nguin | Call Centric's dirt cheap DIDs are $2.95/mo. How'd you get a discount on a dirt cheap DID? |
| 02:56.17 | SeRi | I am not sure. when I went thre there check out I was charged less. |
| 02:56.28 | SeRi | one sec. |
| 02:56.35 | SeRi | is looking at the bill |
| 02:57.28 | SeRi | p3nguin, It was even cheaper. lol by 2c |
| 02:57.36 | SeRi | Nov 7, 2011Products purchasedBalance$-2.26$0.00$-2.26DID - Dirt Cheap DID - |
| 02:57.43 | p3nguin | They don't have a dirt cheap DID in my area code, or I probably would have bought one back in the beginning when I was trying to figure out who I was going to use. |
| 02:58.07 | SeRi | I see. |
| 02:58.32 | SeRi | I dont use it. I just thought it was cheap as hell to use it as a back up. |
| 02:58.35 | p3nguin | Before I even had a DID with anyone, I used CC for internal calling. |
| 02:58.45 | SeRi | ooo cool. |
| 02:59.40 | SeRi | Well thats how I was attracted to them I wanted to make free calls to PR. Than you told me voip.ms does the same. So I stayed at voip.ms and I am glad I did |
| 03:00.08 | p3nguin | You don't even need any ITSP at all to make calls between phones. |
| 03:01.11 | SeRi | yea you mentioned that in the past. the issue we came across was port 5060... though is all fix now. |
| 03:01.43 | SeRi | I could theoretically use asterisk to make peer calls now. |
| 03:01.43 | p3nguin | I assume you could have worked around that blockage with a sip proxy... but you never got around to it. |
| 03:02.38 | SeRi | True. I guess I just got to impatience. Well that was not my only purpose. I also wanted out of the damn 45.99 comcast tel bill. |
| 03:03.06 | SeRi | I sort of did. not quite though due to their lame "bundle" bs |
| 03:04.27 | SeRi | If I get internet "alacarte" its like 200.00 dollars. |
| 03:04.54 | SeRi | just for the internet |
| 03:06.01 | SeRi | with phone the deal drops to like 150.00 |
| 03:06.37 | p3nguin | The whole bundle is $50 cheaper than internet only? |
| 03:07.07 | SeRi | Yes Sr. |
| 03:07.11 | SeRi | I dont get it. |
| 03:07.28 | p3nguin | So that's why you kept your Comcast phone number, I take it. |
| 03:07.34 | SeRi | Yes sr. |
| 03:08.00 | p3nguin | But then you added the cost of DIDs and termination with an ITSP. |
| 03:08.26 | SeRi | lol. |
| 03:08.33 | p3nguin | Does not compute. |
| 03:08.44 | SeRi | I am still cheaper. like 30 dollars. |
| 03:08.56 | SeRi | I was paying 230.00 a month |
| 03:09.28 | SeRi | maybe a bit more after fees and all. |
| 03:09.30 | p3nguin | Maybe you're using Verizon math or something, but where I come from we don't pay $230 for $150 service plan. |
| 03:10.00 | SeRi | no that was because I used there tv as well. "triple bundle" |
| 03:10.11 | SeRi | I drop tv. |
| 03:10.16 | p3nguin | That kicks it up another $80? |
| 03:10.46 | SeRi | HD Complete XF Triple Play "229.99" |
| 03:10.52 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 03:11.08 | SeRi | check it out your self :) Is not my math. is theirs. |
| 03:11.36 | SeRi | So I drop to only local phone and the extream pkg. |
| 03:11.48 | p3nguin | Now you're down to $150. |
| 03:12.12 | SeRi | I pay around 110.00 for internet and about 24.95 for local phone only. |
| 03:12.38 | p3nguin | Okay, so you're paying for phone service with Comcast to have a bundle rate... |
| 03:12.47 | p3nguin | But then you pay for DIDs and termination elsewhere. |
| 03:12.56 | p3nguin | Why not use the phone you're paying for with Comcast? |
| 03:13.11 | SeRi | If you split the two they go well over two hundred dollars. which I dont understand why. |
| 03:13.49 | SeRi | Because of problems with them. My line has bean nothing but hell. plus I refuse to pay for 50.00 doallrs a month for tel service. |
| 03:15.01 | p3nguin | I understand using another carrier due to problems. |
| 03:15.13 | SeRi | The long distance rates are way cheaper in any itsp than it would be with comcast |
| 03:15.26 | p3nguin | They make you pay for calls?! |
| 03:15.39 | p3nguin | Wow. I'm not familiar with that concept. |
| 03:15.44 | SeRi | Yes sr! |
| 03:15.50 | p3nguin | Not familiar. |
| 03:15.54 | SeRi | Calls to MX? |
| 03:16.13 | p3nguin | Oh, international calling. I understand being charged for THAT. |
| 03:16.19 | SeRi | You have to pay an arm an a leg for it. |
| 03:16.21 | p3nguin | I thought you were saying US calls. |
| 03:16.25 | SeRi | oo no |
| 03:16.27 | SeRi | sorry |
| 03:17.06 | SeRi | My wife is from MX. plus I have to be following my brother via tel where ever he travels :/ |
| 03:17.33 | p3nguin | So basically you need to get your SIP credentials from Comcast and put your Comcast "line" on your Asterisk box, and call US numbers on the Comcast plan that you already pay for, and use your ITSP for international calling or SIP URI calling. |
| 03:18.26 | p3nguin | Well, I worded the end of that incorrectly. |
| 03:18.29 | SeRi | lol. Thats funny. I actually tried that and I got "sorry sr we do not use conventional VoIP service. We have our own technology" I was like wtf? |
| 03:18.41 | SeRi | which could be true |
| 03:18.44 | p3nguin | Use your ITSP for international calls, or just use SIP URI. |
| 03:18.47 | SeRi | but I am sure its still voip |
| 03:18.51 | p3nguin | Nah, they're lying. |
| 03:19.04 | p3nguin | I'd bet they use SIP. |
| 03:19.28 | SeRi | I wish I could sniff after the modem :/ |
| 03:20.44 | SeRi | I see the point you are trying to make. I guess I am fed up with them I am sure other would do the same. |
| 03:20.58 | SeRi | as I* |
| 03:21.03 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 03:21.07 | SeRi | not to mention I am tide to a contract. |
| 03:21.12 | SeRi | :( |
| 03:21.34 | SeRi | So Ill ride it out till the contract runs out. |
| 03:22.48 | p3nguin | Explore the Solarwinds software. It finds a lot of useful information on a cable network. |
| 03:22.56 | SeRi | any how.... It wouldn't be fun if dont use asterisk. :) |
| 03:23.01 | p3nguin | You'll just have to connect your computer right to the modem to do it. |
| 03:23.01 | SeRi | I* |
| 03:23.20 | SeRi | Solarwinds. I use some of their software before |
| 03:27.16 | SeRi | Interesting application. I guess this what they did to clone modems. |
| 03:28.17 | p3nguin | It can be useful to gather info, but it doesn't clone a modem. |
| 03:29.18 | SeRi | I know. But they used it along with other tools to clone modems |
| 03:29.29 | SeRi | they would gather all the info they need it |
| 03:30.12 | SeRi | useful app indeed. Thanks. |
| 03:31.46 | p3nguin | There are also other tools to gather info on a cable network. |
| 03:31.59 | SeRi | cmt? |
| 03:32.17 | *** join/#asterisk cbwest (~cbwest@rrcs-70-63-158-29.midsouth.biz.rr.com) |
| 03:32.21 | SeRi | There was one called cmt for motorola modems |
| 03:32.28 | SeRi | I used that one back in the days |
| 03:32.30 | SeRi | cmt diag |
| 03:32.31 | p3nguin | Never heard of it. |
| 03:32.39 | SeRi | or something along those lines |
| 03:32.50 | p3nguin | Right this moment, I don't remember the names of the ones I'm thinking about. |
| 03:32.55 | p3nguin | Coax Thief, maybe? |
| 03:33.05 | p3nguin | I'd have to go look it up. |
| 03:33.41 | SeRi | Thats it |
| 03:33.44 | SeRi | Coax Thief |
| 03:33.55 | SeRi | ~jfgit |
| 03:34.12 | SeRi | lol well I just googled it :) |
| 03:34.51 | p3nguin | I have routers on any modem I have, so most of those things aren't too helpful for me... so I don't retain the names so much. |
| 03:35.17 | SeRi | same here.... |
| 03:35.26 | SeRi | routers that is* |
| 03:37.12 | SeRi | conf in* |
| 03:38.20 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 03:38.23 | cbwest | open question - is there a planned release date for asterisk 10? The repos are empty for now...http://packages.asterisk.org/centos/5/asterisk-10/x86_64/RPMS/ |
| 03:38.51 | p3nguin | We don't plan release dates around here. When it's done, it'll be released. |
| 03:39.15 | cbwest | fair enough…what is the burn list down to? |
| 03:39.24 | cbwest | I saw RC2 released today |
| 03:39.28 | cbwest | I'm hopeful :) |
| 03:40.06 | SeRi | p3nguin, if your up to it I am at the chann* |
| 03:40.14 | SeRi | for it* |
| 03:40.30 | SeRi | damn my English is worst than ever. |
| 03:43.10 | *** part/#asterisk cbwest (~cbwest@rrcs-70-63-158-29.midsouth.biz.rr.com) |
| 03:43.54 | *** join/#asterisk cbwest (~cbwest@rrcs-70-63-158-29.midsouth.biz.rr.com) |
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| 04:05.49 | SeRi | p3nguin, I have not have any issues since rdesktop. |
| 04:05.50 | SeRi | Thanks |
| 04:14.20 | p3nguin | alias rdesktop='rdesktop -g1024x768 -a16 -xb -rsound:local:alsa -rclipboard:PRIMARYCLIPBOARD' |
| 04:14.35 | hardwire | rawr? |
| 04:16.47 | SeRi | :) |
| 04:16.54 | SeRi | nice p3nguin |
| 04:27.01 | *** join/#asterisk MaliutaLap (~biteme@202.124.75.251) |
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| 05:18.40 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 05:27.16 | SeRi | It never hurts to remember those days. |
| 05:27.16 | SeRi | http://video.google.com/videoplay?docid=-6746139755329108302 |
| 05:32.18 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 05:38.23 | SeRi | p3nguin, is it possible to use a did as means fo more channels? ie: if I want to have two simultaneous calls but my provider only allows me two open channels at the time is it possible that asterisk could move the second session to a another provider? |
| 05:39.17 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 05:41.21 | p3nguin | It would be fairly hard to have one DID on more than one provider. |
| 05:42.24 | WIMPy | At least they tell you so. |
| 05:43.17 | p3nguin | And two channels provides two calls, so there's no problem in that respect. |
| 05:43.32 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xjbtsqyugsygdmtk) |
| 05:43.44 | SeRi | p3nguin, well I guess I use a wrong example |
| 05:43.58 | p3nguin | It seems typical for an "unlimited" DID to allow two channels (inbound calls). |
| 05:44.15 | p3nguin | Per minute DIDs give a lot more channels. |
| 05:44.23 | SeRi | 25 at voip.ms |
| 05:44.46 | p3nguin | That's 25 inbound calls at once. Per DID. |
| 05:45.17 | SeRi | I see. I guess the best solution is to have a second pap2 tide to another phone and have that phone dial out threw another context |
| 05:45.35 | p3nguin | Why? |
| 05:45.50 | p3nguin | Why would you need to have another context just to make a call from another device? |
| 05:45.55 | SeRi | so if I have a conference goin with two people that I called that tides my two channels correct? |
| 05:46.28 | p3nguin | Dialing OUT has nothing to do with DIDs and inbound channels. |
| 05:47.03 | p3nguin | You should be able to make at least 25 outbound calls at one time. |
| 05:47.35 | p3nguin | Even if you have the unlimited DID, which is restricted to two inbound channels, you still pay per minute per channel on outbound calls. |
| 05:47.51 | SeRi | ah I see. I thought the limitation was set with outbound calls. |
| 05:47.59 | p3nguin | And DIDs have nothing to do with outbound calls. Period. |
| 05:48.21 | p3nguin | I used VoIP.ms for a year for termination without ever having a DID with them. |
| 05:49.06 | p3nguin | Then one day I decided I wanted to be able to receive calls too. |
| 05:49.09 | SeRi | understood. did are only for incoming . |
| 05:49.46 | p3nguin | If you ever hit a limit on outbound channels, they'll give you more if you need them. |
| 05:50.05 | SeRi | I am sure I am not. I was just confused. |
| 05:50.21 | p3nguin | Unlimited DIDs = two inbound channels per DID |
| 05:50.31 | SeRi | got it. |
| 05:51.40 | p3nguin | And with that, I'm off. |
| 05:51.47 | SeRi | g/n |
| 05:51.52 | SeRi | Thanks for the help |
| 05:54.22 | WIMPy | Sleep. I knew I missed something... |
| 05:56.10 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-254.lv.lv.cox.net) |
| 05:58.01 | Besticles | Good evening. I am having problems with Asterisk properly detecting DTMF tones from an T1 to Analog. I've tryed relaxdtmf=yes, i have boosted rxgain, and despite google reporting that I should turn echo cancellation off, I am finding ec actually helping. The hit rate seems to be 80%. Is there anything else that I can do to improve? |
| 05:59.51 | WIMPy | Play with RxGain. |
| 06:00.49 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-254.lv.lv.cox.net) |
| 06:00.50 | *** join/#asterisk jkroon (~jkroon@dsl-241-225-254.telkomadsl.co.za) |
| 06:03.17 | ChannelZ | looks like he has more than DTMF problems |
| 06:03.41 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 06:04.40 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 06:12.12 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 06:17.31 | Sakuranbo | help |
| 06:18.51 | SeRi | call 911 or 900 or 800 |
| 06:19.06 | Sakuranbo | sorry, wrong message |
| 06:19.08 | SeRi | 999 even |
| 06:19.32 | Sakuranbo | yeah, here 999 works |
| 06:20.06 | SeRi | :) |
| 06:20.30 | irroot | has "exten => 666,1,System(reboot) |
| 06:21.00 | SeRi | irroot, rofl! nice!!!! |
| 06:22.58 | irroot | it comes in handy sometimes |
| 06:23.12 | SeRi | Man I can agree |
| 06:23.19 | SeRi | I didnt know you could do that |
| 06:23.46 | irroot | if you not running as root need to suid or the like |
| 06:24.04 | SeRi | I see |
| 06:26.43 | irroot | also new app_queue @ +/- 30 calls with recording and all the trimings is running at a load of 0.1 where it previously was 0.4 |
| 06:27.07 | irroot | but suspect as the calls back up it will go up |
| 06:29.35 | SeRi | .1 is nice. |
| 06:31.24 | SeRi | specially with all those calls |
| 06:31.46 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 06:33.38 | irroot | its a nice box has 150+ polycom phones and bunch of soft phones |
| 06:34.06 | irroot | good test bed for the rework of app_queue im busy with |
| 06:38.07 | *** join/#asterisk godmachine-x6 (~godmachin@funtoo/user/godmachine-x6) |
| 06:39.42 | SeRi | cool |
| 06:40.36 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 06:54.02 | *** part/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 06:54.08 | LiuYan | ~dynamic realtime |
| 06:54.36 | LiuYan | ~realtime |
| 06:54.36 | infobot | rumour has it, realtime is a feature of Asterisk starting with 1.2 which allows you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
| 06:55.13 | irroot | LiuYan hi there what up |
| 06:57.04 | LiuYan | hi irroot, I'm trying dynamic realtime config, I added "sippeers => odbc,asterisk" to extconfig.conf, renamed sip.conf/users.conf to xx.old, then SIP module can't loaded now |
| 06:58.41 | irroot | LiuYan ah you still need a sip.conf unless you pulling the configs from realtime |
| 06:59.00 | irroot | sippeers will only look at the "peers" not the [genral] |
| 07:01.04 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:01.31 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:05.21 | LiuYan | If I get sip.conf back, then sippeers are readed from sip.conf file, not from database, I followed http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html#dynamic_realtime, the table is not created yet, and I did not got any "column missing" warnings |
| 07:06.33 | irroot | LiuYan take the peers out of the sip.conf you can have both realtime and static peers |
| 07:08.09 | irroot | also ensure the odbc config module is loaded res_config_* |
| 07:16.24 | LiuYan | hmmm, res_config_odbc.so failed to load |
| 07:16.26 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
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| 07:37.54 | LiuYan | have to downgrade to official unixODBC 2.2.11 to get res_config_odbc.so working |
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| 07:49.31 | *** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158) |
| 07:50.10 | RiceCracker | test |
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| 07:56.16 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:56.18 | schmidts | good morning |
| 07:57.01 | ollii | morning |
| 07:57.50 | RiceCracker | morning, testing completed, thanks :) |
| 07:58.59 | Sakuranbo | RiceCracker: Heeelloo |
| 08:06.48 | *** join/#asterisk qakhan (~qakhan@182.185.137.111) |
| 08:07.35 | qakhan | hi all |
| 08:07.53 | ChannelZ | What's that all about? |
| 08:08.04 | qakhan | can anyone plz tell me about branded / multi call center concept |
| 08:08.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:10.13 | qakhan | channelz can you help me? |
| 08:10.36 | ChannelZ | Not really sure what you're asking |
| 08:10.56 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:11.31 | ollii | call center for multiple companys? |
| 08:12.02 | qakhan | yes |
| 08:12.11 | ChannelZ | Yah.. I still don't understand what the question is |
| 08:12.47 | qakhan | tell me about branded / multi call center concept |
| 08:12.50 | ollii | <3 |
| 08:12.54 | ChannelZ | They exist. |
| 08:13.02 | ollii | It is possible. |
| 08:13.09 | qakhan | one agent can answer for multipul companies |
| 08:13.19 | ollii | he can |
| 08:13.47 | qakhan | how? |
| 08:13.51 | ollii | number <-> queue <-> agent ... to make life for agent easier, set callerid(name) = company name |
| 08:14.27 | ollii | exten => 123,1,Set(CALLERID(name)=Company A) |
| 08:14.27 | ollii | exten => 123,n,Queue(Company_A) |
| 08:14.27 | ollii | exten => 123,n,Hangup() |
| 08:14.37 | ollii | exten => 456,1,Set(CALLERID(name)=Company B).... |
| 08:14.53 | ollii | ~book |
| 08:14.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 08:15.20 | ollii | Agent A is in Queue Company_A and Company_B |
| 08:15.22 | irroot | ollii i like using a 2/3 digit billing code that the agents know of there is also ability to play agent a file before connect |
| 08:16.00 | ollii | that also sounds good |
| 08:19.52 | qakhan | how agent come to know which company's call is coming in |
| 08:19.56 | qakhan | ? |
| 08:20.19 | ChannelZ | As olli set, said the CallerID name to the company |
| 08:20.51 | ChannelZ | So when the agent's phone rings, he sees what company it's for. Or use Jabber or something to send them all kinds of information to their computer as the call rings. |
| 08:20.55 | ollii | that might be the easiest version...callerid(name) might not be persisent enough (ldap lookup could overwrite it) |
| 08:21.08 | ollii | or like irroot said |
| 08:22.24 | ChannelZ | Or if they have multiline phones... there's a few ways to do it |
| 08:22.48 | qakhan | ok plz let me know these ways |
| 08:23.13 | schmidts | question to all in here, what do you use for billing? i mean the normal cdrs in the master files, a database (specially what kind of db) or something else? |
| 08:23.29 | ChannelZ | Well you coulld write a custom program to scroll it on an LED message board that sits above the agent's monitor |
| 08:23.31 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 08:23.45 | ChannelZ | Or have it yell out the company over a loudspeaker |
| 08:23.53 | ChannelZ | Or print out on a dot matrix printer |
| 08:24.31 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 08:24.31 | bulkorok | schmidts: i'd use cdrs in db and make a custom "frontend" or integrate in others... |
| 08:24.51 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:25.24 | ollii | account code or "dst" in cdr |
| 08:25.38 | bulkorok | schmidts: but I would also replicate the db on a second server, so that queries don't make any load on the phone-server |
| 08:31.18 | schmidts | bulkorok i though about using my mysql cluster for this, or better postgres |
| 08:31.41 | bulkorok | I've used mysql... |
| 08:32.20 | wdoekes2 | schmidts: cdr_odbc, and then a script that converts those into usable records |
| 08:32.49 | wdoekes2 | (conversion takes place on a slave mysql) |
| 08:33.11 | bulkorok | yeah something like taht... for sake you can make your own table-structure and define in asterisk what comes in... |
| 08:33.12 | *** join/#asterisk qakhan (~qakhan@182.185.167.3) |
| 08:33.32 | qakhan | sorry channelz got disconnected |
| 08:33.33 | ollii | i would prefer an external mysql cluster as a cdr backend...if there's such a call volume which provides noticeable load |
| 08:33.36 | qakhan | plz tell me |
| 08:33.52 | wdoekes2 | bulkorok: yeah.. but I have lots of local dials.. so I get more than one cdr for calls |
| 08:34.05 | ChannelZ | qakhan: My point was that you can do almost anything, I don't know what would work for your particular setup. |
| 08:35.36 | ollii | it sounds like you should contact an * consultant |
| 08:36.02 | bulkorok | wdoekes2: just use cdr_adaptive_odbc and make special account-codes in the dialplan. cdr_adaptive_odbc can filter... |
| 08:36.06 | ChannelZ | You could do anything from custom ringtones on the phones if they supported that, sending agents IMs with information, to just playing a sound file to the agent before the call is connected |
| 08:36.06 | bulkorok | filter accountcode => somename |
| 08:36.23 | ollii | thats why i love * |
| 08:36.52 | bulkorok | wdoekes2: filter src => 123 |
| 08:38.38 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 08:38.55 | tompaw | Good morning. |
| 08:39.16 | ChannelZ | goodnight actually |
| 08:39.19 | ChannelZ | wanders off to bed |
| 08:40.23 | tompaw | :-) Guys, I'm trying to get the ID of an agent that picks up a queue call. Wiki says it should be stored in ${MEMBERINTERFACE}. The problem is, after Queue() call, the DP is not executed. |
| 08:40.37 | tompaw | So my question is - WHERE do I read the ${MEMBERINTERFACE} value from? |
| 08:40.45 | irroot | wdoekes2 schmidts rather use cdr_odbc_adaptive :P |
| 08:41.06 | *** join/#asterisk Tim_Toady (~moi@188.4.58.138.dsl.dyn.forthnet.gr) |
| 08:41.13 | irroot | tompaw run a macro on connect |
| 08:42.29 | tompaw | irroot: can I pass an external value to that macro somehow? I guess you're talking about [,macro from Queue() call, right? |
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| 08:44.33 | kaldemar | tompaw: which wiki mentions MEMBERINTERFACE? |
| 08:44.36 | irroot | tompaw indeed it will run on connect you can then do waht you want therer |
| 08:44.56 | tompaw | http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 08:45.33 | schmidts | irroot i will think about odbc ;) |
| 08:45.49 | kaldemar | tompaw: do you have setinterfacevar=yes? |
| 08:45.56 | tompaw | kaldemar: yes. |
| 08:47.00 | kaldemar | tompaw: where are you trying to get the value? show your configs. |
| 08:47.07 | irroot | or use function QUEUE_VARIABLES |
| 08:47.29 | irroot | to get other stats |
| 08:47.45 | tompaw | What I want sounds simple, but is not so: upon incoming call I want to 1) assign it to an Agent, 2) put both the caller and the Agent in unique confbridge, 3) get the ${EXTEN}-${MEMBERINTERFACE} pair. |
| 08:48.10 | tompaw | kaldemar: sure, let me clean them up a little and pb. |
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| 08:48.58 | irroot | tompaw queue macro is your friend |
| 08:49.47 | irroot | any particular reason for this ?? who else is in conf bridge ? |
| 08:52.01 | tompaw | because I want to be able for the agent to invite different parties to the confbridge during the call |
| 08:52.33 | tompaw | I was planning to use applicationmap for that purpose, but failed miserably due to lack of ${EXTEN} (which is also ConfBridge ID) in their "layer" |
| 08:52.38 | irroot | i use the queue connect macro to fix up call groups start recording [outside of queue as there extra requriements for it] and some other cleanups |
| 08:52.55 | tompaw | so I thought I will just make a web-based dashboard with ajax-based buttons "invite1" "invite2" etc |
| 08:53.08 | tompaw | so ultimately my system just needs to store the ${EXTEN}-Agent pair |
| 08:53.16 | tompaw | anyway, here's my DP so far: http://pastecode.com/eF |
| 08:53.19 | irroot | in the macro you will have both "channels" |
| 08:55.23 | tompaw | Does this make any sense at all? |
| 08:55.35 | tompaw | irroot: if yes, then great, I'm going to test now. |
| 08:57.13 | tompaw | So in line 8 I just add a macro to the Queue, right? What would be the variable names for both channels? |
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| 08:58.35 | tompaw | (inside the macro I mean) |
| 08:58.48 | irroot | yip then when the agent picks up that macr?? |
| 08:59.03 | irroot | is run |
| 08:59.44 | irroot | before the agent and channel are bridged |
| 09:01.32 | tompaw | irroot: please have a look now: http://pastecode.com/eG |
| 09:01.46 | tompaw | Now ${MEMBERINTERFACE} works :-) |
| 09:02.01 | tompaw | But how do I get the Extension value inside that qinfo macro? |
| 09:03.43 | tompaw | It says -- Executing [s@macro-qinfo:1] |
| 09:03.56 | tompaw | Which means the extension is not passed through I'm afraid :-( |
| 09:04.31 | irroot | you can set it as a variable on channel or read info from CDR .... |
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| 09:06.28 | tompaw | irroot: variable? so can I pass it as an argument: exten => _XXX.,1,Queue(SSEN,,,,,,qinfo(${EXTEN})) ? |
| 09:06.42 | irroot | no you can set it |
| 09:07.01 | irroot | using set may need _ |
| 09:07.19 | tompaw | You mean right before calling Queue? |
| 09:07.27 | irroot | SET(_MyInfoForLater=XXXX) |
| 09:07.31 | irroot | yip |
| 09:07.51 | irroot | but it may not be copied across |
| 09:08.08 | tompaw | According to the docs, it should be. Lemme test. |
| 09:09.22 | tompaw | BINGO! |
| 09:10.12 | tompaw | irroot: working version: http://pastecode.com/eH |
| 09:10.17 | tompaw | Thanks for your time :-) |
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| 10:21.57 | LiuYan | cdr_adaptive_odbc failed execute "INSERT INTO cdr (..., end, ...) VALUES(...)" SQL on PostgreSQL 9.1.1: syntax error near 'end', how to fix this? |
| 10:25.05 | LiuYan | ~cdr |
| 10:25.06 | infobot | cdr is probably Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
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| 10:43.18 | _N1x | hi guys , im testing asterisk with sipp tools , how many concurrent calls is possible in 1.8 version? |
| 10:43.32 | _N1x | i have hardware but when i'm generating RTP channels |
| 10:43.44 | _N1x | my server is verry slow but i see hardware |
| 10:43.46 | _N1x | not is full loaded |
| 10:44.06 | _N1x | cpu and memory also |
| 10:44.40 | wdoekes2 | LiuYan: end is a reserved keyword, it should be quoted |
| 10:45.06 | wdoekes2 | but I'm not sure asterisk will do that for you |
| 10:45.24 | wdoekes2 | easiest fix is to rename your column :P |
| 10:45.39 | LiuYan | wdoekes2: thanks, but the SQL statement is/maybe generated by cdr_adaptive_odbc, we can't control it |
| 10:45.54 | _N1x | schmidts: are you here bro? :) |
| 10:46.10 | ollii | _N1x: check your ulimits |
| 10:46.21 | _N1x | i have ulimit -n 65000 |
| 10:46.31 | ollii | ulimit -a |
| 10:46.34 | _N1x | when i'm generating only |
| 10:46.38 | ollii | ah ok |
| 10:46.40 | _N1x | SIP Channels without |
| 10:46.42 | _N1x | RTP |
| 10:46.47 | _N1x | everything is ok and i have |
| 10:46.51 | _N1x | 1000 conncurent calls |
| 10:47.01 | _N1x | but with RTP .... max 200 |
| 10:47.07 | ollii | what are you using to create a rtp stream? |
| 10:47.20 | _N1x | ollii: sipp , copy full command? |
| 10:47.54 | _N1x | sipp -sn uac REMOTEIP -s 2000 -d 1000000 -l 1000 -mp 5062 -mi MYIP - for rtp |
| 10:48.29 | _N1x | and for only SIP without mp and mi options |
| 10:48.40 | LiuYan | wdoekes2: I've just "alias end => CallEndTime", module reload cdr_adaptive_odbc.so, but it still use end as column name |
| 10:48.41 | kaldemar | _N1x: asterisk does not limit the amount of calls it handles unless you tell it to. |
| 10:49.31 | _N1x | kaldemar: so what can i do? or what happening |
| 10:49.46 | _N1x | if i have good hardware but |
| 10:49.53 | _N1x | performance is bad... |
| 10:49.55 | ollii | nothing to see in debug / verbose output on your asterisk cli? |
| 10:50.34 | kaldemar | _N1x: maybe your network is the bottleneck. |
| 10:55.04 | ollii | _N1x: try to use sipp on your asterisk server then you might see if its your network |
| 10:56.41 | LiuYan | wdoekes2: hmmm, I must alter column names of cdr table BEFORE "module reload cdr_adaptive_odbc.so", so that alias take effect now. thanks! |
| 10:57.33 | _N1x | ollii: hm 10x i'll test |
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| 11:07.00 | _N1x | ollii: how i can see |
| 11:07.11 | _N1x | rtp connection in asterisk server? |
| 11:07.16 | _N1x | i'm sniffing with wireshark from |
| 11:07.18 | _N1x | remote network |
| 11:07.23 | _N1x | but not in current server |
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| 11:18.08 | _N1x | guys |
| 11:18.16 | _N1x | how to see if i have RTP Connection |
| 11:18.17 | _N1x | ? |
| 11:18.18 | _N1x | not sip |
| 11:18.24 | _N1x | in asterisk |
| 11:25.55 | teloniusz | there is no such thing as rtp connection, only rtp packets |
| 11:26.53 | teloniusz | from CLI: rtp debug ip <ip> |
| 11:27.00 | *** join/#asterisk willcooke (~will@canonical/willcooke) |
| 11:27.03 | teloniusz | then see if anything shows up |
| 11:28.44 | willcooke | Hi - I have a retro style phone connected to a tdm400 card. I'd like to make the dial tone suitably retro as well. I know the frequencies I want, but I can't work out how to change the dial tone. Is it even possible? |
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| 11:50.54 | _N1x | teloniusz: rtp set debug ip 127.0.0.1:5262 |
| 11:50.57 | _N1x | im doing like this |
| 11:51.00 | _N1x | but can't see |
| 11:51.07 | _N1x | when im turning on rtp set debug on |
| 11:51.11 | _N1x | isee rtp packets is sent |
| 11:51.23 | _N1x | but when im turning off rtp packet and turn on only sip |
| 11:51.26 | _N1x | i see also |
| 11:51.29 | _N1x | rtp packet sents |
| 11:51.38 | kaldemar | willcooke: select a zone with DAHDI system.conf parameters loadzone and defaultzone. |
| 11:53.22 | kaldemar | willcooke: if you want to make your own, add a zone to zonedata.c in dahdi-tools. |
| 11:53.57 | willcooke | kaldemar, I suspected that I was going to have to do that. Ok, well I'll give it a go. Thanks a lot! |
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| 12:12.20 | ollii | _N1x: with using wireshark you should also see rtp streams |
| 12:12.36 | ollii | i might be wrong but theres also a filter for rtp in wireshark |
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| 12:13.32 | _N1x | ollii: from cli? |
| 12:13.50 | ollii | tcpdump -ni eth0 -w foo.pcap |
| 12:13.58 | ollii | import that pcap file into wireshark |
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| 12:37.11 | irroot | hi folks any one got a cure for stupidity that does not include a long prison sentance / death penalty |
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| 12:56.29 | ollii | irroot: sorry, no |
| 12:57.14 | ollii | but you castigate you |
| 12:57.18 | ollii | +could |
| 12:57.55 | singler | irroot: if noone will find out, there will be no punishment :) |
| 12:58.18 | ollii | thats the wrong way...he noticed his own fault...so he has to torture himself |
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| 12:59.03 | singler | as I understand he want to cure someone else |
| 12:59.20 | ollii | then he should torture this person |
| 12:59.29 | singler | 'that does not include a long prison sentance / death penalty" |
| 12:59.35 | singler | :) |
| 13:00.24 | WIMPy | I think the world wide accepted way to deal with stupidity in a legal way is promotion. |
| 13:00.42 | ollii | singler: yeah...torture near death? ;o |
| 13:02.12 | irroot | singler tourture without sentance is only for ex G7/8 presidents and there consorts |
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| 13:24.56 | webon | Fax in Asterisk running. Asterisk accepts faxes ans stores the tif file, but does not go further after ReceiveFAX function , so I am unable to do anything with the received file after the fax has been saved. |
| 13:24.58 | webon | CLI log when receiving fax: http://pastebin.com/c8RtagbE |
| 13:25.00 | webon | extensions.conf extract: http://pastebin.com/0GRXicSt |
| 13:25.02 | webon | Any help? |
| 13:25.30 | file | once the fax is received the call is generally hung up |
| 13:25.42 | [TK]D-Fender | hangup = call goes to "h" |
| 13:25.43 | file | you need to execute your logic in the 'h' extension which is executed when the call is hung up |
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| 13:30.29 | irroot | webon the System application is usefull you can pass it the tiff file name i have a script that converts to PDF and emails a person |
| 13:30.43 | irroot | could share the folder with samba for example |
| 13:31.22 | irroot | a default public shared folder is most useful "virtual fax tray" |
| 13:31.54 | webon | well, all I want to do is to convert to pdf and email the file |
| 13:32.53 | irroot | webon if you nice i may share my script |
| 13:33.04 | SeRi | g/m all. |
| 13:33.35 | SeRi | irroot, you script it outside of *? |
| 13:33.48 | irroot | indeed |
| 13:34.05 | SeRi | I do the same. Is good to know I am not the only one |
| 13:34.08 | SeRi | :) |
| 13:34.30 | SeRi | convert tif to pdf and than email |
| 13:34.36 | irroot | little bash script uuenview / tiff2pdf |
| 13:34.44 | SeRi | :D |
| 13:35.03 | webon | tiff2pdf and mutt |
| 13:40.52 | webon | iroot, great if you want to share your script. Now I'm testing sending two System calls on dialplan |
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| 13:44.08 | iprouteth0 | good morning folks |
| 13:44.22 | iprouteth0 | need some help with asterisk |
| 13:44.49 | iprouteth0 | getting error "cause 16" and "cause 47" |
| 13:45.15 | [TK]D-Fender | Google up "ISDN cause #" |
| 13:45.27 | proute | Hello, |
| 13:45.36 | iprouteth0 | some pri calls fail to connect. not every call fails, the higher the number of active pri channels, the more likely the call will to complete |
| 13:45.44 | bulkorok | iprouteth0: http://networking.ringofsaturn.com/Routers/isdncausecodes.php |
| 13:45.57 | proute | I use tcpdump to check a voip call and I save this check in a pcpap file. |
| 13:46.03 | irroot | pastebin.com/UBKPtcrg webon |
| 13:46.13 | tompaw | [TK]D-Fender: hello, not sure if you remember our conversation from yesterday about adding new participants to existing confbridge, but that's how I worked it out: http://pastecode.com/eK |
| 13:46.22 | tompaw | basically, the key was Set(_VARIABLE....) |
| 13:46.37 | tompaw | as it propagates down to applicationmap macros. |
| 13:47.29 | WIMPy | iprouteth0: Do you have all channels available for outgoing calls? Do you have incomming calls as well? |
| 13:47.49 | proute | So, do you think when I execute and kill tcpdump command, that hangup a call ("break" the voip call)? |
| 13:47.59 | proute | thank for your explaination |
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| 13:48.30 | [TK]D-Fender | proute, No. tcpdump isn't your call, it's just a packet capture. |
| 13:48.30 | webon | BTW, thank you guys! fax is working like a charm |
| 13:49.35 | [TK]D-Fender | tompaw, I don't see where you tak the input of the person to invite in there... |
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| 13:50.48 | iprouteth0 | wimpy: yes. |
| 13:50.57 | tompaw | [TK]D-Fender: these persons are going to be static extensions, as defined in macros macro-nway-invite1 and macro-nway-invite2 |
| 13:51.36 | [TK]D-Fender | tomI thought your goal was to invite otehr non-fixed people in... |
| 13:51.40 | tompaw | in one case it's going to be a pstn number, in another a queue on another asterisk box, in both cases from our perspective it's just a static sip url |
| 13:51.53 | WIMPy | iprouteth0: Yes what? And how do you dial out? |
| 13:52.06 | tompaw | [TK]D-Fender: nono, the idea was for the conference room to be unique to an ${EXTEN} and then being able to invite separate parties independantly from each other. |
| 13:52.09 | irroot | tompaw greets seems you came right ? connex died earlier |
| 13:52.09 | proute | [TK]D-Fender, ok, but does tcpdump has an impact or event on the ethernet card? |
| 13:52.25 | [TK]D-Fender | proute, No, it is jsut a passive capture |
| 13:53.09 | tompaw | irroot: yes, so far it's working great, only dies when there's a g729+g711 codex mix, for some reasons when I invite another g729 to a conf (as a 3rd party) the only g711 is getting muted (and * cries about codec incompatibility) |
| 13:53.38 | tompaw | as long as there's 3x g729 or 3xg711, no problems at all. |
| 13:53.42 | [TK]D-Fender | proute, If I walk up and listen in on you talking with your neighbour your conversation does not suddenly get disrupted when I decide to walk away. |
| 13:53.53 | [TK]D-Fender | tompaw, Basically you don't have licenses installed. |
| 13:54.04 | [TK]D-Fender | tompaw, And transcoding fails |
| 13:54.14 | SeRi | [TK]D-Fender, excellent analogy. lol |
| 13:54.42 | tompaw | [TK]D-Fender: I do have these free modules from asterisk.hosting.lv |
| 13:54.52 | iprouteth0 | wimpy - yes have all channels available for outgoing calls and do get incoming calls as well. |
| 13:54.53 | tompaw | but anyway, it's going to be 3xalaw, so no sweat. |
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| 13:55.23 | [TK]D-Fender | tompaw, Which nobody here really wants to hear about. In most places that's patent evasion |
| 13:55.44 | [TK]D-Fender | tompaw, Now if it fails.. then you probably didn't install the right one |
| 13:55.47 | tompaw | [TK]D-Fender: right, I would actually like to know once and for all which countries do and which don't incorporate g729 patents. |
| 13:56.17 | tompaw | [TK]D-Fender: g729+g711 conference works great. When I try to invite ANOTHER g729 to the room, then g711 is being muted. |
| 13:56.47 | tompaw | (g711 is the invitor, so he leaves the conference for a moment, maybe the conference realizes both parties use the same codec and switches to a non-transcoding mode or sth like that?) |
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| 13:57.39 | [TK]D-Fender | tompaw, Shouldn't... |
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| 13:58.51 | tompaw | [TK]D-Fender: if you look at my macro, the invitor (g711) first moves the invitee to the conference, at this second there's g729+g729 and it LOOKS like that's when * realizes - hey, let's switch the transcoding off |
| 13:59.08 | tompaw | so a moment later, when g711 wants to join back, it says there's a codec issue |
| 13:59.25 | tompaw | I can actually capture the error, gimme amoment. |
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| 14:02.37 | tompaw | LOL, this time it worked with no error :F |
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| 14:04.55 | tompaw | ok, I don't know what u did but it helped, no error ;> |
| 14:06.18 | [TK]D-Fender | I somehow inspired you to look ... which is something almost nobody ever really does on their own... |
| 14:09.53 | pushpop | I was wondering how would I make asterisk register to flowroute for incoming and outgoing calls and keep all my phones on the lan still register to the pbx on port 5060? |
| 14:10.10 | pushpop | [TK]D-Fender dont say anything |
| 14:10.14 | pushpop | because im just messing with you |
| 14:10.15 | pushpop | haha |
| 14:11.08 | tompaw | pushpop: I feel deja vu ;> |
| 14:11.18 | [TK]D-Fender | Really? Because I could completely believe that you'd be in here for the 4th consecutive day asking the same question and expecting different results... |
| 14:11.32 | tompaw | pushpop: don't you need like 2 pbxes for that? |
| 14:11.37 | tompaw | I think it would be the simplest solution |
| 14:12.08 | pushpop | Ya im working on that today |
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| 14:51.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:06.04 | francisvgarcia | pushpop when asterisk connets to the flowroute server It connect as a client and I doesn't use the 5060 port for registering, It use an dynamic port from the range 1024-65535, |
| 15:06.04 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 15:07.27 | francisvgarcia | but If flowroute is going to send a call to the asterisk box It does it like a client to and connect to the asterisk server at port 5060, this is because the Client-Server schema |
| 15:08.42 | francisvgarcia | so you can have multiple client to server connections at the same time using the same port |
| 15:15.04 | p3nguin | But your asterisk client does register TO port 5060. |
| 15:15.18 | p3nguin | (unless you specify an alternate port) |
| 15:16.12 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 15:16.16 | jacc0 | hi all |
| 15:16.27 | [TK]D-Fender | francisvgarcia, If you missed yesterday's little piece of enlightenment as to the purpose of his request it was his idea of a potential solution to some number of calls getting dropped at some point. Not really a valid looking idae to begin with and not metric provided to understand the scope of the actual issue. |
| 15:16.52 | [TK]D-Fender | p3nguin, as above.. not really any point to this whole deal. |
| 15:18.22 | jacc0 | what about SQL_ESC()? there is documentation about this function but it doesn't seem to be in 1.8.8. is there a replacement in 1.8.8? should I use QUOTE()? (it sounds that is by fare not making it sql-injection safe |
| 15:18.24 | jacc0 | ) |
| 15:18.31 | francisvgarcia | [TK]D-Fendeor ok, I missed the yesterday talk |
| 15:18.44 | p3nguin | tompaw: The typical thing to do is use asterisk on the standard port and use a SIP proxy for the non-standard port. But this isn't the answer to his problem anyway. |
| 15:20.04 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 15:21.30 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 15:24.26 | sysreq | hey all. anyone knows why asterisk won't let you set a 'timert1' or 'timerb' (sip.conf) that's lower than 200ms in 1.8 (vs no minimum in 1.6).. i used to use a combination of T1/B timers that would give me a < 5000 ms timeout on INVITEs between servers on the same LAN (for quick fail-over purposes), now I'm up to 200ms*64 (12800 ms).. |
| 15:25.49 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 15:33.13 | jacc0 | is there any standard function in asterisk to protect from sql injection? |
| 15:34.22 | [TK]D-Fender | jacc0, No. Use AGI for this... |
| 15:34.52 | jacc0 | okay, so I would use a PHP agi with mysql_real_escape_string |
| 15:35.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:35.54 | p3nguin | What would cause MixMonitor() to suddenly begin recording calls' channels out of sync? |
| 15:36.10 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:38.12 | jacc0 | is there any way to make Parameterized Queries in asterisk ? |
| 15:38.43 | [TK]D-Fender | jacc0, Any serious stuff like this is best handled in AGI |
| 15:39.18 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:39.29 | jacc0 | Hmm, okay, guess I'll be making strict input validation; because I want to keep it pure asterisk |
| 15:40.52 | jacc0 | or maybe.......................................I'll look up the mysql_real_escape_string sourcecode from php and creat a function for aasterisk with it |
| 15:42.17 | Katty | throws things |
| 15:42.21 | Katty | shreds the curtains |
| 15:44.20 | jacc0 | is checkingout the PHP svn |
| 15:44.26 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 15:44.35 | treborsux | I use analog 12 port cards with software echo and have no issues |
| 15:44.35 | treborsux | <treborsux> will this be the same for the t1 pri cards |
| 15:44.35 | treborsux | <treborsux> or is echo hardware an absolute for t1 pri card? |
| 15:44.57 | [TK]D-Fender | Nothing is absolute. Your milage may vary. |
| 15:46.01 | mateu | how does asterisk determine the sequential ordering of MOH files? |
| 15:49.49 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 15:49.52 | WIMPy | treborsux: Depends a lot on what yu intend to do with your calls. |
| 15:53.04 | p3nguin | List some conditions. |
| 15:53.53 | Katty | conditions. |
| 15:53.56 | Katty | angry. |
| 15:53.59 | Katty | hungry. |
| 15:54.06 | Katty | exhausted |
| 15:54.11 | Katty | cranky |
| 15:55.13 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
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| 15:57.37 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 15:59.39 | treborsux | PoE on twenty-four 10/100/1000 ports supplies up to 7.5W per port, or on twelve 10/100/1000 ports supplies up to the IEEE 802.3af maximum of 15.4W per port |
| 15:59.52 | treborsux | THink I can hook 24 polycom 501 601s to that? |
| 16:01.37 | p3nguin | You can almost never run 24 PoE devices on a 24-port PoE switch at one time. |
| 16:01.53 | leifmadsen | a discussion we had last week seems to indicate that you can |
| 16:02.21 | leifmadsen | because that 15.4W is peak usage, and the POE switch should be smart enough not to enable all ports at once, so once the phones are up, their draw is very low |
| 16:02.28 | WIMPy | 7.5W/port seems plenty to me |
| 16:03.21 | WIMPy | Err. for huge amounts of "low". |
| 16:06.24 | mateu | WIMPy:, p3nguin, Say I have a MOH channel class that is of mode 'file', then 'moh show files' gives a file list in a deterministic order but I haven't figured out how that order is determined. |
| 16:06.45 | mateu | it's not alphabetic, that much I know. |
| 16:07.14 | WIMPy | Probably just the order in which teh files are listed in the directoy. |
| 16:07.20 | mateu | no |
| 16:07.41 | mateu | not matchng 'ls' anyhow. |
| 16:08.02 | WIMPy | ls usually sorts it's output |
| 16:08.19 | mateu | right alphabetic (case insensitive) |
| 16:08.58 | mateu | so 'moh show files' seems to have a different sort, but I can't tell what yet, maybe another file stat other than name. |
| 16:09.09 | WIMPy | That depends on your locale setting. |
| 16:09.36 | mateu | sure, but I'm just dealing with ASCII in my test case. |
| 16:09.39 | r0m|u | p3nguin, Thanks for the callcentric work around it actually works very well. |
| 16:10.37 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 16:10.42 | mateu | a quirk I'm seeing with 'moh show files' is that the order of play doesn't quite match. If I move the first listed file to the last position the listing then matches the play order. |
| 16:10.52 | WIMPy | The "case insensitive" part is controlled by the locale as well. |
| 16:11.06 | timeshell | leifmadsen Need further assistance with ASTERISK-18685 |
| 16:11.24 | WIMPy | only ever used ramdom order. |
| 16:11.34 | mateu | WIMPy: I dont' care about ls sorting, but rather the sorting of the sequential playback. |
| 16:12.13 | timeshell | David Woolley says "You don't seem to have included the verbose logging". Is this done in a way other than setting SIP debug to 10 and enabling dumphistory? |
| 16:12.57 | timeshell | Additionally, my biggest concern about the comments is that we didn't start experiencing the issues before upgrading from 1.6.x to 1.8.x |
| 16:13.24 | *** join/#asterisk jkroon (~jkroon@41.53.248.167) |
| 16:13.43 | timeshell | So something between 1.6.x and 1.8.x must have changed in the way it's dealing with these. |
| 16:14.08 | p3nguin | r0m|u: I... don't... remember any workarounds for it. I only remember configuring asterisk according to CallCentric's topology. |
| 16:14.38 | carrar | pokes Katty |
| 16:14.52 | WIMPy | mateu: From the musiconhold.conf: "Files are played in normal sorting order (same as a sorted directory listing)" |
| 16:15.17 | mateu | WIMPy: yeah, I've found that statement to not be true ;( |
| 16:15.28 | WIMPy | So if that's not the case, maybe you're using different locales for your user and for Asterisk or there's a bug. |
| 16:16.09 | mateu | i'm on older (1.6.x) so I may be stuck with it. |
| 16:16.11 | catphish | lol, whoonearthisTamo |
| 16:16.30 | tompaw | Guys, with ConfBridge, is it posible to disable this "you are currently the only person in the conference" message? |
| 16:16.46 | r0m|u | p3nguin, You add it each CC host in the peer entry |
| 16:16.46 | mateu | WIMPy: thanks for looking into to it some. |
| 16:16.46 | p3nguin | Yes. core show applications ConfBridge |
| 16:16.55 | p3nguin | tompaw: ^ |
| 16:17.10 | p3nguin | r0m|u: Right. That's a correct configuration for their topology. |
| 16:17.56 | p3nguin | s/ir to/ir stupid to/ |
| 16:18.22 | r0m|u | now that sounds much better |
| 16:18.33 | WIMPy | mateu: 'moh show files' doesn't list them sorted for me, either. |
| 16:19.04 | p3nguin | My moh show files does not match my ls output. |
| 16:20.38 | p3nguin | I still need to know what would cause MixMonitor() to suddenly begin recording calls' channels out of sync. |
| 16:21.07 | p3nguin | It usually only lasts for the first few seconds of the call. After a little bit, the audio matches up. |
| 16:21.13 | r0m|u | p3nguin, I submitted a ticket to them explaining my soltuion to there fucked up topology and they told me I have it wrong that I need it to change it back. I didnt bother with them and just closed the ticket. All I wanted them to be aware is that a current config can not play nice with their suggested settings. |
| 16:21.35 | p3nguin | They're idiots. |
| 16:21.43 | r0m|u | agreed. |
| 16:22.55 | p3nguin | Any ITSP that thinks you should use allowguest=yes for the sole purpose of allowing their systems to call inbound to your system because the peer entry never matches... idiots. |
| 16:23.23 | p3nguin | I'm thinking about calling them up right now and telling them that they are idiots. |
| 16:23.26 | jkroon | represents an ITSP and agrees with p3nguin |
| 16:23.31 | r0m|u | Amen to that. |
| 16:23.46 | r0m|u | p3nguin, recorded if you do! LOL |
| 16:23.52 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:23.54 | r0m|u | record it* |
| 16:24.01 | p3nguin | jkroon: Do you know what's going on? |
| 16:24.53 | jkroon | it's the asses that doesn't how how to do authentication. |
| 16:25.05 | jkroon | they basically does "authentication" based on your contact address. |
| 16:25.27 | jkroon | and expect you to do the same. |
| 16:25.28 | p3nguin | Call Centric sends calls inbound to the end user from nine different IP addresses, but their recommended configuration only accounts for the main IP address, so the calls never match the peer entry. |
| 16:25.44 | jkroon | ooh, ouch, even worse. |
| 16:26.10 | jkroon | anyway, gotta run, always fun to hear that the only idiots aren't in ZA :p |
| 16:26.18 | p3nguin | Their fix: allowguest=yes and general context= some context that allows for all your inbound calls from Call Centric. |
| 16:26.57 | p3nguin | Stupid people. My fix: configuration to match all of their IP addresses. |
| 16:27.12 | *** join/#asterisk grEvenX (~even@pc194-213.ktv.no) |
| 16:27.36 | tompaw | p3nguin: ;> thx, I have M1q, but it still says the prompt. |
| 16:27.39 | r0m|u | and it works. :) |
| 16:27.40 | WIMPy | Yes, but we really need a way to configure multiple IPs for a peer. |
| 16:28.09 | WIMPy | Manually checking the list of IPs for a given host regularly is rather iffy. |
| 16:28.09 | catphish | that would be nice, but multiple peers never hurt anyone |
| 16:28.36 | WIMPy | No, but the list of IPs can change any time. |
| 16:28.52 | catphish | so can the list of peers :) |
| 16:29.06 | catphish | but it is a pain certainly |
| 16:29.46 | p3nguin | wimpy: My solution is simple: http://pastebin.com/X0eHWYA2 |
| 16:29.55 | WIMPy | Well, the thing is that you won't know until you find out that you don;t accept the calls any more or worse, only sometimes. |
| 16:30.55 | p3nguin | If you are concerned that the IP address could change, be sure to use the host name. And pray that they have working DNS. |
| 16:31.03 | WIMPy | Yes, that's the way I do it with UI, butwhen they decide to add more hosts to the list, calling me from them will become a gamble. |
| 16:31.09 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:31.10 | wcselby | o/ |
| 16:31.52 | WIMPy | So if you want to be sure it works, you'd have to check the list regularly. |
| 16:32.29 | p3nguin | I think the established providers don't do things like that as much. It's the new ones that fiddle with things regularly. |
| 16:32.38 | WIMPy | well, if you're lucky and they hace a DNS name that points to all relevant IPs that is. |
| 16:33.02 | WIMPy | They regularly add new nodes. |
| 16:34.14 | *** join/#asterisk mmcji (~mmcji@65.172.54.254) |
| 16:34.29 | treborsux | anyone have a suggestion of a lower priced 24 port poe I am looking at the SRW2024P |
| 16:34.43 | r0m|u | treborsux, Good luck with that |
| 16:35.02 | treborsux | do you suggest 2 12s? |
| 16:35.17 | r0m|u | treborsux, they are usually not cheap and if you need 24 ports most likely would be a 42 port switch. |
| 16:35.27 | r0m|u | most switches do not do all ports PoE |
| 16:35.41 | r0m|u | any how look in to zyxel |
| 16:35.53 | wcselby | treborsux what is your definition of a "lower priced"? |
| 16:36.09 | r0m|u | hola wcselby |
| 16:36.09 | wcselby | r0m|u I'm seeing more and more that do it on all 24 ports |
| 16:36.18 | wcselby | of course, the price range is like 600+ |
| 16:36.23 | treborsux | not over a grand |
| 16:36.23 | r0m|u | ^^ |
| 16:36.41 | r0m|u | wow they have come a long way since I looked at them. |
| 16:37.02 | r0m|u | a 24port poe use to be more than a grand and they where not all 24 ports. |
| 16:37.17 | wcselby | the cisco SRW224G4P-K9-NA supposedly does poe on all 24 ports |
| 16:37.54 | wcselby | from a local houston supplier (you can probably find cheaper online somewhere) - http://www.altex.com/Cisco-24-port-POE-Ethernet-Switch-SRW224G4P-K9-NA-P149949.aspx |
| 16:38.39 | wcselby | that one is 589 from altex |
| 16:38.45 | wcselby | i'm sure you can get it cheaper somewhere online |
| 16:40.31 | r0m|u | not bad. |
| 16:40.34 | r0m|u | I stand corrected |
| 16:43.12 | r0m|u | p3nguin, can I give you credit for the correct setting on CC? |
| 16:43.15 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:43.45 | p3nguin | *shrug* |
| 16:43.48 | r0m|u | lol |
| 16:44.00 | p3nguin | I guess that depends on what you're trying to do. |
| 16:44.04 | r0m|u | They re open the ticket. a "senior tech" |
| 16:44.15 | r0m|u | he ask who came up with it. |
| 16:44.22 | r0m|u | I wan to say you |
| 16:44.28 | r0m|u | with your permission of course |
| 16:44.31 | eppigy | hello |
| 16:44.32 | eppigy | i am dave |
| 16:44.47 | p3nguin | Tell him that your consultant did, when the calls were not working. |
| 16:44.57 | r0m|u | go it. |
| 16:45.02 | r0m|u | got it* |
| 16:47.31 | p3nguin | I kind of blame Asterisk for this being the resulting configuration. If SRV was followed on all calls, it shouldn't be a problem. |
| 16:47.55 | p3nguin | I've never been able to see that SRV works correctly, though. |
| 16:50.08 | r0m|u | I see. |
| 16:51.00 | r0m|u | lets see what they say now. |
| 16:51.42 | p3nguin | In light of Asterisk's inability to work fully with SRV, they should be providing GOOD configuration samples for Asterisk. |
| 16:52.31 | p3nguin | Just for the record, I have never seen a single ITSP that has provided Good configuration samples for Asterisk. |
| 16:52.57 | p3nguin | Much of the time, the samples don't even work at all. |
| 16:53.13 | p3nguin | Other times, it's a hack... like CallCentric provided. |
| 16:53.19 | carrar | Why should the ITSP have to configure your server? |
| 16:53.29 | p3nguin | They shouldn't HAVE TO. |
| 16:53.38 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:53.55 | p3nguin | But when they say the support Asterisk configuration and they provide sample configurations that you are supposed to use, the samples should be reasonable. |
| 16:54.13 | carrar | Asterisk customers may be the lowest $ of customers |
| 16:54.16 | carrar | err % |
| 16:54.24 | r0m|u | I dont think is a matter of them doing it but to at least supply the correct examples for a broken env such as CC |
| 16:54.27 | eppigy | 1%'ers |
| 16:54.32 | carrar | heh |
| 16:54.36 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:54.49 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-391586.home.otenet.gr) |
| 16:55.05 | carrar | So they need to setup a ASterisk Lab with all the changing version to keep current? |
| 16:55.17 | p3nguin | A complete moron should be able to sign up for the service, copy the Asterisk samples, paste into the relevant files, adjust user-specific data, and have a working system. |
| 16:55.31 | [TK]D-Fender | carrar, YES. |
| 16:55.38 | r0m|u | carrar, have you look back and see why we are talking about this? |
| 16:55.41 | p3nguin | Much of the time, this does not happen. |
| 16:55.43 | *** join/#asterisk gruvfunk (18a121cf@gateway/web/freenode/ip.24.161.33.207) |
| 16:55.43 | carrar | How is the "complete moron" gonna setup asterisk? |
| 16:55.53 | gruvfunk | greets all |
| 16:56.00 | carrar | r0m|u, I have not :) |
| 16:56.04 | r0m|u | carrar, I called them robots. copypasta and magically works! |
| 16:56.04 | [TK]D-Fender | carrar, Um.... we see them in here all the time ;) |
| 16:56.05 | p3nguin | Haven't you seen the types we get in here? |
| 16:56.07 | WIMPy | carrar: By joining #asterisk |
| 16:56.08 | carrar | I just like randomly interject |
| 16:56.15 | [TK]D-Fender | high-5's p3nguin |
| 16:56.25 | r0m|u | rofl!!!!!!!!! |
| 16:56.45 | carrar | Maybe that "complete moron" can read the book and configure his own peer!! :) |
| 16:56.55 | p3nguin | No. He can't read. |
| 16:56.58 | p3nguin | Apparently, anyway. |
| 16:57.05 | carrar | heh |
| 16:57.14 | r0m|u | lol |
| 16:57.23 | gruvfunk | building 1.8.7 on Amazon EC2 AMI release 2011.09 -- but make config does not work to install the init script... which contrib/init script should I use on this centos-like system? I tried redhat didn't work. |
| 16:58.41 | p3nguin | I guess what we need to do is index the entire book in infobot. That way when we tell someone to read the book for the answer, they aren't so intimidated by and entire book. Having a problem configuring sip.conf? ~book-sip.conf |
| 16:59.07 | r0m|u | lol..... |
| 16:59.13 | p3nguin | Need voicemail to work? ~book-voicemail.conf |
| 16:59.24 | r0m|u | well at least have it make a snip of it and point the page number |
| 16:59.35 | p3nguin | MoH not playing the files you thought you had? ~book-musiconhold.conf |
| 16:59.42 | r0m|u | LOL!!!! |
| 16:59.45 | r0m|u | ~moron |
| 16:59.46 | infobot | from memory, moron is someone that types long lines starting with "infobot" |
| 16:59.56 | p3nguin | hahahaha |
| 17:00.02 | p3nguin | I've seen a few of those. |
| 17:00.03 | r0m|u | rofl!!!!!!!! |
| 17:00.09 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 17:00.25 | gruvfunk | anyone here build using Amazon's AMI? |
| 17:00.45 | p3nguin | We occasionally get someone who tries to have conversations with infobot. |
| 17:01.02 | r0m|u | hahahahahaha no way! |
| 17:01.19 | p3nguin | How do you think that response about a moron came about? |
| 17:01.42 | eppigy | hello i am a complete moron |
| 17:01.48 | r0m|u | hahahahahahahah omfg! |
| 17:02.08 | p3nguin | infobot: Do you like sausage? |
| 17:02.08 | infobot | yes, I like sausage. |
| 17:02.25 | WIMPy | Anyone volunteer for an incomplete moron *eg* |
| 17:02.25 | eppigy | infobot: are you sausage? |
| 17:02.29 | p3nguin | infobot: Do you know how to configure Asterisk? |
| 17:02.29 | infobot | yes, I know how to configure Asterisk. |
| 17:02.54 | r0m|u | LMAO |
| 17:02.57 | r0m|u | it talks! |
| 17:03.00 | r0m|u | LOL |
| 17:03.03 | p3nguin | infobot: Are you willing to tell others how to configure Asterisk? |
| 17:03.08 | [TK]D-Fender | <eppigy> hello i am a complete moron <- no ... you are Dave! |
| 17:03.26 | [TK]D-Fender | ~areyouadog ? |
| 17:03.26 | infobot | Bark! Bark! |
| 17:03.29 | [TK]D-Fender | GOOD BOY! |
| 17:03.33 | r0m|u | LMAO |
| 17:03.34 | [TK]D-Fender | ~botsnack |
| 17:03.34 | infobot | aw, gee, [TK]D-Fender |
| 17:03.57 | eppigy | hello |
| 17:03.58 | eppigy | i am dave |
| 17:04.04 | [TK]D-Fender | this is factual |
| 17:04.09 | eppigy | FACT |
| 17:04.28 | r0m|u | ~factual |
| 17:04.32 | r0m|u | :/ |
| 17:04.38 | r0m|u | lol |
| 17:05.10 | gruvfunk | glad to see so much banter in here, but seriously - anyone building Asterisk on Amazon's default AMI? which init script should I use? |
| 17:05.52 | *** join/#asterisk kb1_kanobe (~kris@bdr2.fieldrd.scrd.ca) |
| 17:06.36 | p3nguin | You know why we haven't answered the question the other dozen times you asked it? Because we don't meet your criteria. Can you imagine if all 226 people all told you no at the same time? |
| 17:07.57 | gruvfunk | p3nguin: dozen times? |
| 17:08.07 | wcselby | gruvfunk I think there are asterisk specific vm's precompiled on amazon, if you can find it |
| 17:08.10 | p3nguin | I may have exaggerated. |
| 17:08.19 | p3nguin | You may have asked three or four times. |
| 17:08.23 | gruvfunk | wcselby: looking for custom build out |
| 17:08.29 | gruvfunk | i asked twice brother p3nguin |
| 17:08.33 | gruvfunk | chill |
| 17:08.33 | r0m|u | p3nguin, CC: "we will bring your recommendation to our managers" |
| 17:08.36 | wcselby | is ami different from ec2? |
| 17:08.52 | wcselby | to me, AMI is the asterisk manager interface |
| 17:09.07 | gruvfunk | in this case we are talking Amazon's AMI's |
| 17:09.12 | [TK]D-Fender | To me it's a BIOS maker |
| 17:09.13 | gruvfunk | Amazon Linux AMI release 2011.09 |
| 17:09.19 | [TK]D-Fender | Or a friend |
| 17:09.21 | p3nguin | I count three, but whatever. |
| 17:09.45 | [TK]D-Fender | Or a relatively common female name |
| 17:09.56 | gruvfunk | guv |
| 17:09.56 | Qwell | Why would being a VM make you use a different init script? |
| 17:10.14 | p3nguin | qwell: Oh, qwell, you know why. ;) |
| 17:10.14 | gruvfunk | the OS type |
| 17:10.38 | gruvfunk | Qwell: in this case I assumed a "redhat" derivative init would work |
| 17:11.00 | gruvfunk | but it has not, tried them all (debian, gentoo, slackware, mandriva, redhat...) nothing works to start asterisk |
| 17:11.01 | p3nguin | So what OS is it? |
| 17:11.12 | gruvfunk | Amazon Linux AMI release 2011.09 |
| 17:11.20 | gruvfunk | from Amazon, it looks to be centos based |
| 17:11.24 | p3nguin | What style of init system does it use? |
| 17:11.26 | r0m|u | p3nguin, is a redhat spin. |
| 17:11.34 | *** join/#asterisk JustinCampbell (~justinCam@c-68-81-39-247.hsd1.pa.comcast.net) |
| 17:11.39 | p3nguin | If it is CentOS, a Red Hat script should work. |
| 17:11.42 | gruvfunk | but I tried redhat and mandriva inits, neither work |
| 17:11.50 | [TK]D-Fender | gruvfunk, So "make config" can't tell what you have... if it's a RH base just copy the init script by hand. That's all MAKE is doing anywhay |
| 17:11.53 | _Corey_ | gruvfunk: I've heard a number of people struggle with the micro instances but not with the regular ones |
| 17:11.58 | [TK]D-Fender | anyway* |
| 17:12.12 | gruvfunk | i've manually copied the file too [TK]D-Fender .. no go |
| 17:12.24 | gruvfunk | interesting _Corey_ thanks |
| 17:12.34 | wcselby | gruvfunk can you just simply run the binary and start asterisk? |
| 17:12.37 | [TK]D-Fender | gruvfunk, clarify what you'v done to evaluate the "no go" |
| 17:12.41 | wcselby | let's get down to brass tacks here |
| 17:12.47 | r0m|u | _Corey_, when recording a call how do I know I am inside my legal boundaries? |
| 17:12.56 | gruvfunk | wcselby: yes |
| 17:12.58 | wcselby | r0m|u speak with a lawyer |
| 17:12.58 | r0m|u | ~faith |
| 17:12.58 | infobot | Telephony is not faith-based. Look. Always look, and then show. |
| 17:13.04 | _Corey_ | r0m|u: Ask a lawyer ;) |
| 17:13.07 | r0m|u | lol |
| 17:13.17 | wcselby | r0m|u but I'm pretty sure Texas is a one-party consent state for recording |
| 17:13.24 | [TK]D-Fender | wcselby, http://verydemotivational.files.wordpress.com/2010/04/129166850360476757.jpg |
| 17:13.29 | gruvfunk | [TK]D-Fender: copying init scripts from source contrib into /etc/init.d and testing them gives errors of different types |
| 17:13.30 | _Corey_ | r0m|u: http://en.wikipedia.org/wiki/Telephone_recording_laws |
| 17:13.41 | wcselby | so as long as one of the parties is notified that at the call may be recorded, you should be good |
| 17:13.43 | [TK]D-Fender | gruvfunk, Feel like showing us? |
| 17:14.01 | r0m|u | Thanks guys I just thought I ask since _Corey_ told me I just couldnt record all calls.... |
| 17:14.04 | wcselby | [TK]D-Fender LOL! |
| 17:14.35 | wcselby | r0m|u I used to consult for a financial firm that recorded any call into particular call centers |
| 17:14.43 | _Corey_ | r0m|u: "This call may be recorded for quality and training purposes" |
| 17:14.45 | wcselby | and I don't even know if they played that message |
| 17:14.47 | p3nguin | I record all calls. I'm stationed in a state that allows it. |
| 17:15.01 | p3nguin | It is regular practice. |
| 17:15.26 | wcselby | r0m|u and this place was inside Houston. but if you want to be completely safe, use _Corey_ 's example. i think there may even be an allison recordin gof htat message in the extra sounds |
| 17:15.27 | gruvfunk | [TK]D-Fender: thank you... that did it... providing you with the show clarified the error on my side |
| 17:15.36 | gruvfunk | rc.redhat.asterisk works fine |
| 17:15.40 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:15.40 | p3nguin | Inbound calls are provided notification that the call is recorded. |
| 17:15.46 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-154-7.chyn.qwest.net) |
| 17:15.47 | gruvfunk | had to update the Asterisk SBIN dir in the script |
| 17:15.53 | [TK]D-Fender | ~faith |
| 17:15.53 | infobot | Telephony is not faith-based. Look. Always look, and then show. |
| 17:15.55 | [TK]D-Fender | ^^^^ |
| 17:15.56 | p3nguin | Outbound calls are recorded as a regular practice. |
| 17:15.58 | r0m|u | Thank You All! |
| 17:16.33 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:16.51 | r0m|u | so for inbound inside a conference I can have an announcement when the user joins that this "can be recorded" |
| 17:17.16 | r0m|u | or as _Corey_ stated : "This call may be recorded for quality and training purposes" |
| 17:17.28 | wcselby | r0m|u just use a Playback() before even entering the Conference() |
| 17:17.30 | wcselby | or whatever |
| 17:17.40 | r0m|u | wcselby, yes thats what I was thinking |
| 17:17.49 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-154-7.chyn.qwest.net) |
| 17:17.53 | wcselby | i hate dealing wth phone systems that aren't under my direct control |
| 17:18.00 | r0m|u | p3nguin, in outbound you dont have to let the other user know you are recording? |
| 17:18.18 | p3nguin | I should, but Illinois is a gray area state for recording. |
| 17:18.26 | r0m|u | ah! nice. |
| 17:18.26 | wcselby | r0m|u not if he's in a state that allows one-party notification |
| 17:18.43 | r0m|u | cool. Texas is one-party. |
| 17:18.46 | r0m|u | nice! |
| 17:18.56 | p3nguin | So you don't have to notify anyone. |
| 17:19.06 | p3nguin | If you're on the call, and you consent, that's all you need. |
| 17:19.07 | r0m|u | I will use it with comcast |
| 17:19.09 | _Corey_ | r0m|u: I start the Monitor() on the incoming call and then play the "this call may be..." announcement. This way it's on the recording that the caller heard it |
| 17:19.27 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:19.28 | p3nguin | Same here. |
| 17:19.43 | r0m|u | _Corey_, thanks for the suggestion! I am going to start working on it now |
| 17:19.54 | r0m|u | fires up nx |
| 17:20.56 | p3nguin | It's a simple three lines: |
| 17:21.02 | p3nguin | exten => s,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes); |
| 17:21.02 | p3nguin | exten => s,n,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.WAV,a); |
| 17:21.03 | p3nguin | exten => s,n,Playback(silence/1&this-call-may-be-monitored-or-recorded&silence/1); |
| 17:21.20 | citywok | * has locked up again but this time i had debug threads enabled |
| 17:21.21 | r0m|u | sweet!!!!!!!! |
| 17:21.25 | citywok | is there anything else i shoudl get before killing it? |
| 17:21.29 | citywok | core show locks/threads |
| 17:21.30 | r0m|u | Thanks p3nguin! |
| 17:21.48 | p3nguin | Just change your exten => s,n to same => n |
| 17:22.03 | r0m|u | yes sr. |
| 17:22.56 | p3nguin | Just place the lines before you do any actual talking, such as through a Dial(). |
| 17:23.25 | r0m|u | damn you readmy mind. I was going to ask that next lol |
| 17:24.35 | r0m|u | pb coming up |
| 17:24.41 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:25.40 | r0m|u | p3nguin, http://pastebin.com/KfTA5Le3 |
| 17:26.34 | p3nguin | Now you can delete line 9. |
| 17:26.42 | p3nguin | Playback provides the Answer. |
| 17:26.48 | r0m|u | ah cool! |
| 17:27.01 | r0m|u | done :) |
| 17:27.22 | citywok | can anybody spot the problem, why * hung up and stopped processing new calls? Threads: http://pastebin.com/wmRDzBht -- Locks: http://pastebin.com/h43XadHp |
| 17:34.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:34.41 | r0m|u | p3nguin, where would it drop the file by default? |
| 17:34.56 | p3nguin | /var/lib/asterisk/monitor/ |
| 17:34.56 | r0m|u | found it |
| 17:34.57 | p3nguin | wait |
| 17:35.03 | p3nguin | /var/spool/asterisk/monitor/ |
| 17:35.06 | r0m|u | found it* :) |
| 17:35.10 | r0m|u | Yes Thanks! |
| 17:35.22 | wcselby | citywok open a ticket on issues.asterisk.org/jira |
| 17:35.28 | wcselby | or whatever the issue tracker is now |
| 17:35.43 | citywok | but i'm scared of leifmadsen - i don't want to get body slammed again |
| 17:35.50 | r0m|u | lmao |
| 17:36.12 | p3nguin | Yay, Thanksgiving potluck over at the office today. I hope people bring food that's worthy. |
| 17:36.16 | wcselby | he shouldn't be bodyslamminig you unless you've done something wrong |
| 17:36.39 | r0m|u | p3nguin, sounds like fun. |
| 17:36.48 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 17:36.55 | wcselby | i work with only two other people |
| 17:37.01 | wcselby | neither of whom are married |
| 17:37.06 | wcselby | i doubt we'll be doing anything like that |
| 17:37.12 | wcselby | they may try to go to a bar sometime next week |
| 17:37.13 | wcselby | but I doubt it |
| 17:37.46 | p3nguin | My wife made some pumpkin pies last night, but she still needs to whip the cream today. |
| 17:38.17 | eppigy | there is a dirty joke in there somewhere |
| 17:38.36 | p3nguin | She used my recipe for the pies, which uses canned pumpkin and canned candied yams. |
| 17:38.48 | r0m|u | drools! |
| 17:38.52 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 17:39.13 | p3nguin | Most people never know it includes candied yams. They just know it's good. |
| 17:39.22 | r0m|u | lol |
| 17:39.41 | citywok | lol wcselby, i'll have him look at the logs BEFORE i post it. lol. |
| 17:39.47 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 17:39.59 | r0m|u | p3nguin, looks like CC took it seriously. they left the ticket open and will follow up with me |
| 17:40.17 | p3nguin | Good. Did you explain to them why I did what I did? |
| 17:40.19 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 17:40.20 | r0m|u | wonders of CC lurks this realms... |
| 17:40.26 | wcselby | p3nguin i can't stand candied yams, I'd be interested to see if I could tell |
| 17:40.27 | p3nguin | I doubt it. |
| 17:40.50 | p3nguin | If you can't stand the taste, you may be able to detect it. |
| 17:41.13 | wcselby | i dont' know what it is, but I've never liked them |
| 17:41.15 | r0m|u | p3nguin, yes I explain why you made the changes. |
| 17:41.26 | wcselby | i'm the only one in my family who doesn't like them |
| 17:42.03 | r0m|u | wcselby, you need one of this http://en.wikipedia.org/wiki/File:Kiekie_yams.jpg |
| 17:48.27 | r0m|u | can somebody engrave into the bot some sort of silly answer regarding the multiple devices and ports? |
| 17:52.35 | *** join/#asterisk nix8n82 (~hmg@75-174-154-7.chyn.qwest.net) |
| 17:53.40 | p3nguin | The answer may be too complex for a one-liner from the bot. |
| 17:53.52 | r0m|u | I see. |
| 17:55.24 | r0m|u | I am not sure why I still have them in my sip.conf: Registration for 'UID@sip2.remacservices.net' timed out, trying again (Attempt #422) |
| 17:55.53 | *** join/#asterisk ruied (~ruied@po-217-129-252-25.netvisao.pt) |
| 17:56.01 | p3nguin | Maybe you're ever hopeful. |
| 17:56.57 | r0m|u | I learn a lot about hope and faith in this channel. |
| 17:57.17 | [TK]D-Fender | There're the last 2 things you lose here... |
| 17:57.23 | [TK]D-Fender | tey're* |
| 17:57.24 | [TK]D-Fender | gah |
| 17:57.26 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 17:57.30 | [TK]D-Fender | they're** |
| 17:57.32 | r0m|u | lol |
| 17:57.32 | [TK]D-Fender | alsjkdhklajsdhapuiywpeuirsdfsdf |
| 17:57.35 | r0m|u | lmao |
| 17:58.21 | *** join/#asterisk zurdo13 (~Jose@201.130.0.10) |
| 17:58.44 | lordvadr | can anybody explain the new DAHDI channel numbering to me? Used to be DAHDI/g0/<chan>, now I get something like DAHDI/i1/-8a, even though I Dial(DAHDI/g0). This is after an upgrade from 1.6.2.11 to 1.8.7.0. |
| 18:00.16 | zurdo13 | hi i am new to asterisk, I have a question dadhi the modules are really needed |
| 18:00.19 | WIMPy | lordvadr: It uses the callerID now. |
| 18:00.40 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-avprfebufqhotydv) |
| 18:01.17 | WIMPy | zurdo13: Are you trying to ask if you need dahdi? |
| 18:01.25 | zurdo13 | ye |
| 18:01.27 | zurdo13 | yes |
| 18:01.35 | *** join/#asterisk frawd (~francois@97.Red-213-98-249.dynamicIP.rima-tde.net) |
| 18:01.41 | lordvadr | WIMPy: That doesn't look like caller ID to me. I'm trying to find out if a channel is available, and which channel it is on a given PRI. ChanIsAvail(DAHDI/go) returned DAHDI/i1/-8a. What does -8a mean? |
| 18:02.01 | zurdo13 | tengo entendedido que sirve para cierto tipo de hardware |
| 18:02.26 | WIMPy | lordvadr: I think that's the call reference. |
| 18:02.41 | zurdo13 | I have used to understand certain types of hardware |
| 18:03.07 | WIMPy | zurdo13: You need dahdi if you have hardware that uses it or if you want to use MeetMe() or Page(). |
| 18:03.18 | lordvadr | WIMPy: Ok, let me ask you this a little differently. I need to know if the call I'm about to place is going to be above channel X. How would I find that out? |
| 18:03.21 | [TK]D-Fender | or IAX2 trunk mode |
| 18:03.48 | WIMPy | lordvadr: You don't |
| 18:03.51 | [TK]D-Fender | lordvadr, the channel name won't tell you what the next one will be |
| 18:05.41 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-mfkwqslvnnpglark) |
| 18:05.56 | shido6 | anyone using an iphone to scan mac addresses from barcodes? |
| 18:06.18 | shido6 | alphanumeric bar scanner |
| 18:07.04 | lordvadr | I did this once before, and it was pretty easy. ChanIsAvail(DAHDI/g0) and I got DAHDI/g0/16 or such |
| 18:07.12 | lordvadr | That was back in 1.6 |
| 18:07.15 | p3nguin | Not iPhone specifically, but I'd imagine it works if you have a scanning app. |
| 18:07.39 | shido6 | i found it |
| 18:07.43 | shido6 | 29.99 for On Hand |
| 18:07.50 | [TK]D-Fender | sad |
| 18:07.50 | shido6 | the others do numbers only |
| 18:08.09 | shido6 | grr... |
| 18:08.17 | WIMPy | lordvadr: That's not a reliable information. The channel can change from when yu check the next free until it's used. And even then it can change again. |
| 18:08.18 | shido6 | I restract my statement |
| 18:08.35 | *** join/#asterisk irroot (~gregory@41.55.56.103) |
| 18:09.17 | WIMPy | lordvadr: Why do you want to know anyway? |
| 18:09.18 | lordvadr | WIMPy: Let me rephrase the question then. I need to know when I hit 75% usage on a PRI. I realize it's not entirely reliable, but it's a good estimate as there's always the race condition. I just need to know when we get up into the high teens reliably. |
| 18:09.48 | WIMPy | lordvadr: Use the GROUP functions. |
| 18:10.16 | edge | Does anybody have a guide to picking phones for Asterisk? |
| 18:10.20 | lordvadr | I have a circuit that is the "primary" circuit in a two pri trunk-rollover group, so all my inbound calls come in on that circuit. I want to use it for outbound until it get's high enough, and then send the calls to the other server that has the "idle" pri. |
| 18:10.29 | WIMPy | And ideally that shouldn't work, as it makes sense to seize channels in different order from both sides. |
| 18:11.24 | WIMPy | lordvadr: Why don't you use the ide link in the first place? |
| 18:11.29 | WIMPy | idle |
| 18:11.44 | lordvadr | WIMPy: It's in a different data center. It's not ideal. |
| 18:12.33 | WIMPy | Ok, so then it's the GROUP functions. |
| 18:13.57 | lordvadr | WIMPy: After a quick read, those look like they will work perfectly. Thank you. |
| 18:16.02 | lordvadr | Next question. We have a sip provider that we send most long distance to. My customers complain about DTMF not being recognized--it's unreliable, one ore more digits get "missed" for lack of a better word. My provider says the SSRC is changing on the DTMF events--it's different than the RTP audio stream's SSRC, and each DTMF event is has a different SSRC. They say this is the problem. I've seen some patches and bugs from a |
| 18:16.02 | lordvadr | year ago that attempted to address this. |
| 18:16.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:16.35 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-qwosenssrhirbiea) |
| 18:16.52 | lordvadr | My question is that which is right? Should the 2833 events use the same SSRC, and why is it broken in 1.6.2.11, 1.8.7.1 and 1.8.6.0. Seems to be a wide range of brokenness. |
| 18:17.23 | lordvadr | All these patches and bugs were back around 1.6.2.7 or so. |
| 18:18.04 | lordvadr | I need to either point them to an RFC that says one way or the other is correct, and I can't find one. I wouldn't imagine asterisk would be this poorly behaved though. |
| 18:19.38 | *** join/#asterisk frawd (~francois@97.Red-213-98-249.dynamicIP.rima-tde.net) |
| 18:26.03 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 18:30.12 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:36.33 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:38.08 | r0m|u | bbl lunch time |
| 18:38.50 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-mfyjeoopoqqycjyj) |
| 18:42.59 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 18:44.03 | Naikrovek | anyone here play minecraft |
| 18:45.50 | *** join/#asterisk irroot (~gregory@197.111.77.204) |
| 18:48.37 | p3nguin | Why would MixMonitor suddenly start recording calls with the two channels out of sync? It starts out not in sync, but usually ends up converging before the end of the call. |
| 18:48.50 | Naikrovek | starts out of sync? |
| 18:49.04 | Naikrovek | weird. mine ended out of sync when there were packet drops, but started out fine. |
| 18:49.30 | p3nguin | So it could be a result of traffic shaping? |
| 18:49.49 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:49.53 | Naikrovek | well mine ENDED out of sync. and one could hear the packet drops on the call |
| 18:49.58 | p3nguin | oh |
| 18:50.25 | Katty | what a mother flippin day i'm having |
| 18:50.38 | eppigy | same |
| 18:50.39 | Naikrovek | switch to tossing. flipping weakens the knees |
| 18:50.49 | p3nguin | Well, the traffic shaper is going to be queueing the call as it goes out, so maybe that actually could be introducing some latency at the beginning. |
| 18:51.05 | Katty | pff. |
| 18:51.11 | Katty | maybe i like weakening knees |
| 18:51.19 | eppigy | oh snapo |
| 18:51.20 | eppigy | snap |
| 18:51.25 | p3nguin | I suppose I can switch off the traffic shaper and see if it goes back to normal. |
| 18:51.28 | Naikrovek | lol "oh snapo" |
| 18:51.29 | Naikrovek | rofl! |
| 18:51.44 | p3nguin | If it does, then I should try a traffic shaper with moar powar. |
| 18:51.52 | Naikrovek | p3nguin: yea try that and see what happens. i'm curious |
| 18:55.32 | p3nguin | I disabled the shaper, and it still is out of sync. |
| 18:56.21 | Naikrovek | eh well it was a longshot anywho |
| 18:56.25 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 18:56.29 | [sr] | howdy people |
| 18:56.37 | p3nguin | It was certainly worth trying. Now I know that's not it. |
| 18:56.38 | [sr] | i'«m looking for something that i cant find |
| 18:57.07 | [sr] | i need a TA that can convert an analog line, to sip, to connect a sip telefone |
| 18:57.27 | [sr] | anyone knows a trademark that has this deviced? |
| 18:58.00 | p3nguin | Linksys/Cisco SPA-3102 |
| 19:00.05 | [sr] | ok it has 1x FXS and 1x FXO |
| 19:00.16 | p3nguin | and one Ethernet |
| 19:00.21 | [sr] | sure :p |
| 19:00.22 | p3nguin | Actually two Ethernet |
| 19:00.35 | p3nguin | But you'll only use one, since the other is a router port. |
| 19:01.18 | [sr] | are you sure it can be configured, to use the sip account of the device, that will use the FXO port? sorry the question :) |
| 19:01.41 | p3nguin | The FXO port connects to your wall jack. |
| 19:02.05 | [sr] | that (in the case will be a tradicional PBX analog line) |
| 19:04.50 | pushpop | Hello p3nguin |
| 19:04.57 | pushpop | I have logs |
| 19:05.08 | pushpop | [TK]D-Fender you there |
| 19:07.21 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-alduujlsawcmudwo) |
| 19:08.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 19:13.25 | [TK]D-Fender | yes |
| 19:13.42 | pushpop | hello, my issues is that some outgoing calls dont go through |
| 19:13.47 | pushpop | and I have sip debug logs |
| 19:13.52 | pushpop | would you mind take a quic look |
| 19:13.58 | pushpop | taking* |
| 19:14.13 | pushpop | we get half a ring then dead air |
| 19:14.22 | pushpop | only happens on some outbound calls others work fine |
| 19:16.31 | *** part/#asterisk kb1_kanobe (~kris@bdr2.fieldrd.scrd.ca) |
| 19:20.27 | r0m|u | pushpop, paste the pb link. dont ask if you could paste it. :) |
| 19:20.59 | *** join/#asterisk jjlee (~john@cpc3-nmal4-0-0-cust774.croy.cable.virginmedia.com) |
| 19:24.12 | pushpop | just have phone numbers in the capture |
| 19:24.16 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 19:24.18 | pushpop | didnt know if that was a good idea to paste |
| 19:24.22 | *** join/#asterisk wesphillips (~wphill04@137.237.195.4) |
| 19:24.44 | *** join/#asterisk irroot (~gregory@197.170.48.114) |
| 19:25.04 | r0m|u | take it out if it concerns you. |
| 19:25.13 | navaismo | pushpop phones, password usernames and so on must be edited before pb |
| 19:25.20 | r0m|u | ^^ |
| 19:25.35 | *** join/#asterisk tessier (~treed@kernel-panic/copilotco) |
| 19:26.07 | r0m|u | ctrl find NUMBER replace WITH bogusnumber |
| 19:26.22 | r0m|u | same for uid and pass |
| 19:28.57 | [TK]D-Fender | We don't care about your phone numbers. Mess with them and you're messing with potential evidence associated with your problem. |
| 19:29.08 | p3nguin | The general rule is "pastebin the entire config, hiding only passwords." |
| 19:29.26 | p3nguin | Note: hiding only passwords |
| 19:29.45 | pushpop | http://pastebin.com/RqC9NtEa is the log and 5552925644 is the number we had issues dialing. |
| 19:30.24 | pushpop | dialed from ext 2103 |
| 19:30.32 | p3nguin | I don't see that you have core verbose set to at least 3 in this pastebin. |
| 19:30.40 | p3nguin | You can't dial "from" extensions. |
| 19:30.57 | p3nguin | You can dial from phones, though. Is your phone named 2103? |
| 19:31.11 | pushpop | 2103 yes |
| 19:31.34 | p3nguin | [Nov 16 13:27:02] WARNING[17638]: pbx.c:5037 __ast_pbx_run: Timeout, but no rule 't' or 'e' in context 'park-dial' |
| 19:31.37 | p3nguin | argh! |
| 19:31.39 | r0m|u | pushpop, the kung fu masters have spoken I shall now stand back. |
| 19:31.45 | r0m|u | p3nguin, another one lol |
| 19:32.07 | p3nguin | How do I create the rule in a context that isn't explicitly defined anywhere? |
| 19:32.13 | pushpop | p3nguin what is that |
| 19:32.23 | p3nguin | I don't have a park-dial context; it's apparently something call parking creates internally. |
| 19:33.05 | jjlee | Perhaps not strictly about asterisk, but definitely a problem faced by this newbie asterisk user: I have an asterisk instance that I'm testing at the moment by calling a sipgate number. asterisk is successfully registered as a SIP client with my sipgate credentials. I'd like to cut out the PSTN middleman and test it using (gasp) SIP instead. My first attempts at doing that with ekiga and linphone failed: ekiga tells me |
| 19:33.05 | jjlee | <PROTECTED> |
| 19:33.23 | [TK]D-Fender | And all I see for that "call" is where * killed it. |
| 19:33.33 | [TK]D-Fender | now try and actually look at the complete call. |
| 19:33.36 | *** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net) |
| 19:34.20 | p3nguin | jjlee: Are you trying to register ekiga to asterisk and call from ekiga to asterisk, and possibly to another device? |
| 19:34.25 | jjlee | no |
| 19:34.46 | pushpop | what more can I do when we call the number it gives us a half of ring then dead air |
| 19:34.56 | jjlee | asterisk is just behaving as a SIP client, so that I can call the sipgate number and have asterisk do voicemail -- that's all I want |
| 19:34.57 | p3nguin | You're trying to register ekiga to sipgate and call your Asterisk via SIP URI from Sipgate? |
| 19:35.12 | jjlee | That works fine, but I just want to test it in a better way than actually calling that sipgate number |
| 19:35.28 | p3nguin | You don't have to use sipgate. |
| 19:35.47 | jjlee | specifically, a way that doesn't involve using my PSTN telephone (since that costs me money) |
| 19:36.01 | p3nguin | Give me a scenario where you'll be using asterisk and not dialing a PSTN number. |
| 19:36.22 | jjlee | p3nguin: when I'm testing asterisk using only SIP, and not a PSTN network |
| 19:36.35 | pushpop | so I guess my log sucked |
| 19:36.39 | p3nguin | You give me the scenario, and I'll tell you how to do it. |
| 19:37.11 | jjlee | p3nguin: I'm not sure specifically where I was unclear |
| 19:37.48 | p3nguin | If you're not willing to cooperate, good luck in your endeavors. |
| 19:38.02 | *** join/#asterisk slidesinger-lt (~jtatum@pool-71-168-246-49.cmdnnj.east.verizon.net) |
| 19:38.07 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 19:38.21 | pushpop | p3nguin me? |
| 19:38.28 | p3nguin | no |
| 19:38.57 | jjlee | p3nguin: essentially, asterisk is functioning as a voicemail system at the moment. I'd like to test that behaviour without using a PSTN phone. That's all, really: I don't think there's anything else to add except for my probably-incorrect guesses at how to accomplish that. |
| 19:39.27 | p3nguin | pushpop: What I want to see from you is core set verbose 3 in addition to the sip debug, at the same time, pastebinned. |
| 19:40.05 | p3nguin | jjlee: How does sipgate come into it, then? Call asterisk from your SIP phone. |
| 19:40.24 | pushpop | p3nguin -vvvr? |
| 19:40.37 | p3nguin | pushpop: "core set verbose 3" for the third time, now. |
| 19:40.48 | pushpop | p3nguin sorry |
| 19:40.57 | p3nguin | jjlee: You don't want to register the SIP phones to Asterisk, but you want to call voice mail via SIP URI? |
| 19:41.11 | jjlee | p3nguin: sipgate comes into it only as explanatory background for the task that I'm trying to accomplish |
| 19:41.25 | jjlee | p3nguin: and as the first SIP address that I tried to call |
| 19:41.26 | p3nguin | So we'll stop talking about it since it isn't related. |
| 19:41.50 | p3nguin | jjlee: I'll ask again: You don't want to register the SIP phones to Asterisk, but you want to call voice mail via SIP URI? |
| 19:42.23 | jjlee | p3nguin: thinks... |
| 19:43.30 | p3nguin | jjlee: If you don't want to register the SIP phones to that Asterisk, you'll have to use a SIP phone that has no problem calling SIP URIs while not registered to *something*. Not all of them do this willingly. |
| 19:43.59 | jjlee | yeah, I guess that's right. My knowledge of SIP is limited to what I've picked up while editing asterisk configuration over the past few hours, so that's half the problem. |
| 19:44.07 | jjlee | (answering your previous question) |
| 19:44.22 | p3nguin | jjlee: Then you'll either need to configure a peer for every possible SIP phone that will ever call Asterisk... or allow all anonymous calls. |
| 19:44.34 | jjlee | re registering: ah, I see |
| 19:46.42 | p3nguin | Registration from a SIP devie to Asterisk simply tells Asterisk how to reach said device. It isn't required to make calls inbound; however, most SIP phones want to be registered to *something* before it will make outbound calls. Some of them even have a setting to allow calling without being registered. To simply use a phone that is not registered to anything and make calls via SIP URI to asterisk, you'll need a phone that ... |
| 19:46.44 | p3nguin | ... supports it (at least has a setting to enable it). |
| 19:48.02 | jjlee | OK, thanks. I will only use SIP for testing. So it appears I just need a free SIP account from some random provider. Then I should be able to call my sipgate SIP URI and get through to my asterisk instance. |
| 19:48.03 | p3nguin | Let us assume you have a phone which will allow you to make calls to SIP URIs without being registered to anything. Just dial the IP address or host name of Asterisk. Sometimes you have to actually tell the phone it is a SIP call by prefixing the IP/hostname with sip: or it will fail. |
| 19:48.34 | [TK]D-Fender | Just register that SIP phone to * like the rest of the world... |
| 19:48.34 | p3nguin | sip:my.asterisk-system.net |
| 19:48.47 | p3nguin | Or actually, I didn't define the extension there... |
| 19:48.57 | p3nguin | sip:voicemail@my.asterisk-system.net |
| 19:49.15 | p3nguin | [tk]d-fender: That's what I'd do, but he apparently has no desire to do that. |
| 19:49.23 | jjlee | p3nguin: right -- that's what I've been doing |
| 19:49.33 | p3nguin | Okay, so... what was the problem again? |
| 19:50.11 | jjlee | as I mentioned, the SIP client "ekiga" failed with "Call end reason for Call... set to EndedBySecurityDenial" in its debug logs |
| 19:50.15 | p3nguin | I understand the scenario. Now I want to know the PROBLEM. |
| 19:50.48 | r0m|u | jjlee, trash ekiga. Its full of bugs |
| 19:50.48 | p3nguin | What have you registered ekiga to during that call/test? |
| 19:50.51 | jjlee | by problem, do you mean what I was expecting to happen? |
| 19:51.07 | p3nguin | No, I know that you're expecting a successful call. |
| 19:51.13 | jjlee | r0m|u: what do you recommend? |
| 19:51.22 | p3nguin | Are you using Windows? |
| 19:51.24 | jjlee | p3nguin: OK. By problem, what do you mean? |
| 19:51.43 | p3nguin | I like zoiper classic for Windows. |
| 19:51.48 | jjlee | p3nguin: ubuntu 10.10 |
| 19:51.54 | r0m|u | ugh. |
| 19:52.03 | p3nguin | I'd use twinkle or I might try out blink. |
| 19:52.40 | jjlee | I'll give those a try, thanks |
| 19:52.51 | p3nguin | By problem, I wanted to know about that failed call. |
| 19:53.06 | jjlee | perhaps the reason that ekiga did not successfully make a call is that it was not registered to any SIP server? |
| 19:53.14 | p3nguin | Could be. |
| 19:53.14 | jjlee | p3nguin: what would you like to know about that failed call? |
| 19:53.28 | p3nguin | I don't think ekiga will make calls without being registered to something. |
| 19:53.44 | r0m|u | ^^ |
| 19:53.45 | jjlee | OK. That sounds like a good place to start digging again! |
| 19:53.56 | jjlee | I'll go away and give those things a try |
| 19:53.57 | jjlee | Cheers |
| 19:53.58 | p3nguin | You could register it to ekiga's service, and then dial your asterisk via SIP URI. |
| 19:54.15 | r0m|u | ekiga CAN NOT make calls without been register. you ether use their service or register to a provider |
| 19:54.16 | jjlee | p3nguin: you might have thought that. In fact, I did think that |
| 19:54.18 | p3nguin | Or just register it to asterisk like everyone else. |
| 19:54.31 | r0m|u | p3nguin, +1 |
| 19:54.32 | jjlee | p3nguin: I was not able to get ekiga to register with ekiga's server :-O |
| 19:54.38 | p3nguin | :/ |
| 19:54.45 | *** join/#asterisk Greenlight (~wluke@cpc4-dund11-2-0-cust378.sgyl.cable.virginmedia.com) |
| 19:54.45 | p3nguin | Register to asterisk. |
| 19:54.45 | jjlee | even though it registered with sipgate fine |
| 19:55.00 | jjlee | perhaps-- another possibly-fruitful idea, thanks |
| 19:55.08 | p3nguin | If you register it to asterisk, you don't even have to worry about SIP URIs at all. |
| 19:55.14 | p3nguin | You'll be able to dial normal extensions. |
| 19:55.28 | jjlee | right |
| 19:55.44 | r0m|u | jjlee, expect ekiga to crash occasionally. Switch while you can |
| 19:55.47 | *** part/#asterisk wesphillips (~wphill04@137.237.195.4) |
| 19:56.34 | p3nguin | I wonder what happens if I define park-dial context in dial plan and provide the t extension in it. |
| 19:56.40 | p3nguin | You think it will break call parking? |
| 19:56.59 | r0m|u | I dont think so. |
| 19:57.14 | *** join/#asterisk ShaunR (~ShaunR@freenode/sponsor/NDChost.com) |
| 19:57.39 | r0m|u | wouldnt it transfer it before it calls on t |
| 19:57.52 | p3nguin | If pbx_config is creating it internally, it could break if I define it explicitly in dial plan. |
| 19:57.57 | pushpop | p3nguin http://pastebin.com/JZCeS9Mm as per your request outbound number 5552925644 |
| 19:58.03 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 19:58.31 | ShaunR | Does asterisk have any extension commands that i can use to write a to a file... basically somthing like noop() that i can just give a filename and have it write it to their. |
| 19:58.31 | r0m|u | p3nguin, I see what you are saying. |
| 19:58.35 | *** part/#asterisk petern_ (~petern@lachesis.fuzzle.org) |
| 19:59.20 | Greenlight | ShaunR: System(echo "test" >> file) ? |
| 19:59.38 | ShaunR | Greenlight: ya, was hoping for somthing alittle better than that. |
| 19:59.50 | Greenlight | WHat is it you need to do ? |
| 20:00.01 | ShaunR | trying to create my own log of whats going on. |
| 20:00.30 | ShaunR | their happen to be a mail cmd or anything? |
| 20:00.44 | r0m|u | ShaunR, use the Verbose application with a redirect |
| 20:00.58 | p3nguin | pushpop: Using good ol' FreePBX, I see. |
| 20:01.04 | pushpop | p3nguin yes |
| 20:01.14 | ShaunR | verbose with a redirect? |
| 20:01.20 | p3nguin | You know this isn't a FreePBX channel, don't you? |
| 20:01.21 | Greenlight | Or have asterisk log everything to a file, and parse that |
| 20:01.29 | p3nguin | And you know I can't support FreePBX, right? |
| 20:01.37 | r0m|u | ~freepbx |
| 20:01.38 | 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, or fpbx |
| 20:01.50 | p3nguin | I'm still looking over the log to see if I see anything blatantly wrong. |
| 20:01.58 | pushpop | p3nguin thank you |
| 20:03.13 | ShaunR | there's a readfile cmd, why not a writefile :) |
| 20:04.03 | [TK]D-Fender | pushpop, [Nov 16 14:52:16] VERBOSE[17067] chan_sip.c: Reliably Transmitting (NAT) to 216.115.69.144:5060: |
| 20:04.03 | [TK]D-Fender | INVITE sip:15552925644@216.115.69.144 SIP/2.0 |
| 20:04.11 | [TK]D-Fender | pushpop, fLOWROUTE IS NOT BEHIND nat. FIX YOUR TRUNK |
| 20:04.47 | pushpop | nat=yes |
| 20:04.52 | pushpop | in peer details |
| 20:04.56 | pushpop | should be removed |
| 20:04.58 | p3nguin | They aren't behind NAT, though. |
| 20:05.23 | [TK]D-Fender | set t NO |
| 20:05.38 | [TK]D-Fender | pushpop, To: <sip:15552925644@216.115.69.144> <-- does flowroute want a 1 in front? |
| 20:06.22 | pushpop | id have to figure that out seems to work everywhere else with the 1 |
| 20:08.22 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:08.51 | [TK]D-Fender | <--- SIP read from UDP:192.168.1.107:5060 ---> |
| 20:08.51 | [TK]D-Fender | CANCEL sip:5552925644@192.168.1.100;user=phone SIP/2.0 |
| 20:08.55 | [TK]D-Fender | Phone cancelled the call |
| 20:09.09 | pushpop | does a hang up cause the cancel? |
| 20:09.24 | [TK]D-Fender | Snom user hung up |
| 20:09.36 | pushpop | yea cause it was dead |
| 20:09.38 | [TK]D-Fender | outbound to flowroute goes down with it |
| 20:09.58 | FLeiXiuS | My SIP peers register fine, but when they begin transmitting after the INVITE, I receive maximum transmission retries errors. |
| 20:10.34 | [TK]D-Fender | FLeiXiuS, PASTEBIN a failed call with SIP DEBUG enabled. |
| 20:10.36 | [TK]D-Fender | ~pb |
| 20:10.36 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:11.42 | FLeiXiuS | Yep |
| 20:12.21 | FLeiXiuS | I second. |
| 20:13.24 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 20:14.26 | r0m|u | ShaunR, I was saying some thing along the lines of exten => 2222,1,Verbose(3,${HAVE THAT REDIRECT IN TO A FILE HERE IE:(>>file)} if is even possible. not sure.... |
| 20:14.41 | autofsckk | hello, i have a spa3102 that it send calls and receive calls, but the thing is that it takes too much time to send the call to the internal extension when a call gets in |
| 20:14.45 | FLeiXiuS | [TK]D-Fender, http://pastie.org/2874074 |
| 20:14.49 | [TK]D-Fender | pushpop, I only say 1 RINGING event from them... |
| 20:15.01 | pushpop | [TK]D-Fender nat=no fixed it |
| 20:15.07 | pushpop | called go through now |
| 20:15.09 | r0m|u | autofsckk, fix your dial plan in your ata |
| 20:15.10 | [TK]D-Fender | \o/ |
| 20:15.10 | pushpop | calls* |
| 20:15.20 | pushpop | [TK]D-Fender thank you very much |
| 20:15.22 | [TK]D-Fender | pushpop, providers are never nat=yes |
| 20:15.28 | pushpop | p3nguin thank you |
| 20:15.35 | pushpop | [TK]D-Fender lesson learned |
| 20:15.40 | [TK]D-Fender | their media could eb from separate servers than their signalling |
| 20:16.02 | pushpop | understood |
| 20:16.14 | r0m|u | autofsckk, go to the webui of your ata and look under "regional" i think is where is at. and fix the dial paln to dial out right away |
| 20:16.26 | r0m|u | is on the admin's manual |
| 20:16.37 | [TK]D-Fender | FLeiXiuS, New call at verbose 10 as well and don't mask #'s |
| 20:17.29 | FLeiXiuS | [TK]D-Fender, I'd rather mask them. This is at verbose 3. |
| 20:17.40 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.152) |
| 20:17.41 | FLeiXiuS | Let me try it again at 10. |
| 20:17.59 | [TK]D-Fender | #'s = evidence and things aren't working. Let me know when I'm not going to waste my time. |
| 20:18.07 | autofsckk | r0m|u: ok let me look for it |
| 20:19.13 | r0m|u | autofsckk, here : http://pastebin.com/02YdVgkQ |
| 20:20.39 | FLeiXiuS | [TK]D-Fender, How is it any different then me leavin the original IP's in there? It's a simple replace all ;-) |
| 20:21.04 | autofsckk | r0m|u: i cant find any dialplan on the regional tab, could it be the pst line? |
| 20:21.34 | autofsckk | or line 1 maybe? |
| 20:22.28 | r0m|u | autofsckk, its under Line1 and Line2 |
| 20:22.50 | autofsckk | r0m|u: ok thanks |
| 20:23.44 | r0m|u | autofsckk, your welcome |
| 20:24.48 | p3nguin | fleixius: You're not going to get more information which is useful by going from 3 to 10, but hiding numbers is bad, m'kay? |
| 20:25.42 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 20:26.43 | autofsckk | r0m|u: is that info for outgoing calls? or for incoming? |
| 20:26.54 | r0m|u | outgoing calls |
| 20:27.02 | r0m|u | there is no such thing for incoming calls |
| 20:27.21 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 20:27.42 | autofsckk | r0m|u: well i have no trouble for outgoing calls, my problem is with the incoming ones, they take too much time to send the call to the extension |
| 20:29.11 | r0m|u | You mean to the phone? |
| 20:29.48 | r0m|u | PB your incoming context. |
| 20:29.57 | autofsckk | r0m|u: yes, i call from an aoutside line to the one connected to the spa, it takes about 4 rings to asterisk show the incoming call in CLI and i then sends the call to one telephone |
| 20:30.22 | r0m|u | PB your incoming context |
| 20:30.44 | autofsckk | http://pastebin.com/0QrNCZC3 |
| 20:31.25 | r0m|u | exten => _5145971,n,Dial(${TEL} <----- thats wrong |
| 20:32.11 | autofsckk | then it isnt my incoming mmm |
| 20:32.22 | autofsckk | wait let me check all the extensions.conf |
| 20:32.24 | r0m|u | exten => _5145971,n,Dial(${TEL}) |
| 20:33.11 | FLeiXiuS | p3nguin, hiding the IP's is bad? How so...I could mask them in any which way so long as its consistent. |
| 20:33.55 | FLeiXiuS | It's a very simple..find this IP and replace it with this one. I could of made it 34.234.53.1 and you would have no idea that I had changed it |
| 20:34.08 | FLeiXiuS | None the less , thats not the problem at hand |
| 20:34.14 | autofsckk | r0m|u: i fix it but i tested it before i fix it and the call worked too, could it be possible? |
| 20:34.19 | r0m|u | autofsckk, I am hoping that your variable is such as SIP/device |
| 20:34.54 | autofsckk | r0m|u: yes it is TEL=SIP/101 |
| 20:35.47 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 20:35.51 | r0m|u | autofsckk, http://pastebin.com/pveAZKZc |
| 20:36.00 | wcselby | ... |
| 20:36.02 | wcselby | testing |
| 20:36.04 | wcselby | is this thing on |
| 20:36.06 | autofsckk | r0m|u: i think the problem is at the spa, because it takes to long to show the incomieng call in asterisk |
| 20:36.38 | r0m|u | in asterisk? |
| 20:36.53 | *** join/#asterisk zurdo13 (~Jose@201.130.0.10) |
| 20:37.09 | r0m|u | asterisk receives the call from the peer and passes along to the device |
| 20:37.17 | r0m|u | asterisk does not take long at all to do that |
| 20:37.28 | r0m|u | maybe the ATA is taking a long time to ring? |
| 20:37.46 | autofsckk | r0m|u: yes, ring or send the call to asterisk |
| 20:38.23 | r0m|u | autofsckk, you are making no sense. You told me the problems where incoming only not both. |
| 20:38.35 | r0m|u | so is it both or just incoming? |
| 20:38.37 | autofsckk | im talking about incoming |
| 20:38.44 | p3nguin | fleixius: It's "could have," and yes, you could have changed it to anything. If you did it consistently, great. The problem is that most people don't do it in a sane manner, so we do not encourage it. |
| 20:38.45 | wcselby | is the spa the incoming line? |
| 20:38.55 | autofsckk | yes |
| 20:39.04 | wcselby | there you go r0m|u confusion solved |
| 20:39.18 | wcselby | some spa's have FXO lines and can be inbound or outbound, not just an ATA |
| 20:39.34 | p3nguin | Which is in this case a 3102. |
| 20:39.37 | wcselby | is it like some spa 3102 or something? |
| 20:40.02 | FLeiXiuS | p3nguin, Most people are idiots ;-) I'm lost. This is for one of my mobile clients whom uses 3G/Wifi. Irony here is that 3G works but wifi doesnt.. go figure. |
| 20:40.40 | r0m|u | wcselby, yes |
| 20:40.44 | r0m|u | it is a 3102 |
| 20:41.22 | *** join/#asterisk xpot (~xpot@66.219.225.48.provo.dynamic.broadweavenetworks.net) |
| 20:41.23 | r0m|u | ok I got it now. the spa IS the incoming line. :) |
| 20:41.45 | autofsckk | r0m|u: yes is the incoming and out too |
| 20:42.20 | autofsckk | p3nguin: the problem is that when i call to the landline from a cell phone, it takes too much time for the spa3102 to send the call to asterisk |
| 20:42.31 | r0m|u | autofsckk, ok than do you have the correct settings for your country under Ring and Call waiting Tone Specs under regional? |
| 20:42.44 | p3nguin | I'd have to see the ENTIRE configuration before I guess what's wrong. |
| 20:42.55 | wcselby | autofsckk I've seen that with a lot of FXO devices |
| 20:43.15 | p3nguin | screenshots of PSTN Line and Line 1, and extensions.conf |
| 20:43.21 | autofsckk | p3nguin: the config from the spa you mean? |
| 20:43.29 | r0m|u | I had a similar issue my problems where my Ring and Tone specs |
| 20:43.37 | p3nguin | My SPA-3102 sends the call in to asterisk straight away. |
| 20:44.19 | autofsckk | p3nguin: ok gimme a sec, im in a asus netbook, it will take some time to screenshotin it ha |
| 20:44.37 | p3nguin | Yeah, I'd guess it will. :/ |
| 20:49.29 | wcselby | windows snipping tool |
| 20:50.14 | wcselby | makes life easy |
| 20:59.36 | autofsckk | p3nguin: http://img4.imageshack.us/img4/790/line1g.jpg |
| 21:04.04 | autofsckk | http://img24.imageshack.us/img24/7240/pstn.jpg |
| 21:05.10 | r0m|u | autofsckk, I think he is going to need more than that. Like SIP settings and Regional. not sure though ill wait and see |
| 21:06.00 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-abbrovbvnqwyvtzk) |
| 21:06.10 | autofsckk | i need to put the extensions.conf |
| 21:06.20 | autofsckk | but i cant c&p :S |
| 21:06.22 | autofsckk | gimme a sec |
| 21:07.39 | autofsckk | http://pastebin.com/zA8GF4pJ |
| 21:07.59 | p3nguin | Your PSTN answer delay is 10 |
| 21:08.01 | p3nguin | Turn it down. |
| 21:09.00 | *** part/#asterisk bobb_WU (~bobb_WU@206.74.211.152) |
| 21:09.05 | autofsckk | p3nguin: done |
| 21:09.08 | autofsckk | i put 2 in there |
| 21:09.41 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-abbrovbvnqwyvtzk) |
| 21:09.47 | p3nguin | And you have two "from-trunk" contexts. Delete one. |
| 21:11.08 | autofsckk | ok |
| 21:11.14 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 21:13.26 | autofsckk | p3nguin: that fixed it thanks |
| 21:13.36 | p3nguin | All better now? |
| 21:14.00 | autofsckk | p3nguin: only one question, if i put 0 in there, it sounds strange before the first tone and then it rings, is it normal? |
| 21:14.14 | p3nguin | I couldn't say because I have never set it to 0. |
| 21:14.33 | p3nguin | I think I use 4 or 6, because I wait for caller ID to be picked up. |
| 21:15.01 | autofsckk | i see |
| 21:15.12 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 21:15.49 | autofsckk | thanks a lot man, again :D |
| 21:16.44 | r0m|u | glad is all worked out |
| 21:17.19 | autofsckk | p3nguin: when i pickup an analog phone plugged to the spa, it sounds very strange at the begining, it doesnt give the normal tone so you can dial, it sounds like 3 short tones and then the normal dial tone, what could it be? |
| 21:17.22 | r0m|u | p3nguin, is the kung fu master |
| 21:17.27 | autofsckk | r0m|u: thanks to you too man |
| 21:17.59 | r0m|u | autofsckk, r0m|u> I had a similar issue my problems where my Ring and Tone specs |
| 21:18.02 | p3nguin | Stutter dial tone? That usually indicates one of your vertical service codes has been activated. |
| 21:18.29 | autofsckk | r0m|u: how did you fixed it? |
| 21:18.41 | p3nguin | The one for voice mail is like 6 stutters, though, not 3. |
| 21:18.47 | autofsckk | i dont know what a stutter tone, let me google it |
| 21:19.14 | autofsckk | yeah its more like 6 short tones |
| 21:19.20 | autofsckk | but im not using voicemail here |
| 21:19.30 | r0m|u | again p3nguin is right ;) |
| 21:20.28 | p3nguin | It's like duh-duh-duh-duh-duh-dial tone |
| 21:20.37 | autofsckk | yes, exactly |
| 21:20.40 | p3nguin | That's MWI. |
| 21:20.50 | p3nguin | Asterisk has told it that it has voice mail, I guess. |
| 21:21.07 | p3nguin | voicemail show users |
| 21:21.23 | autofsckk | i see |
| 21:21.35 | autofsckk | is there a way to shut that ? |
| 21:22.08 | p3nguin | Do you have mailbox=something in your peer entry in sip.conf? |
| 21:23.04 | p3nguin | I guess you can just turn it off on the ATA and forget about it in asterisk. |
| 21:23.08 | autofsckk | nop |
| 21:23.42 | p3nguin | On the Line 1 tab, there is at least one setting for MWI. |
| 21:24.35 | p3nguin | I don't know the difference between VMWI Serv and MWI Serv, though. |
| 21:24.44 | p3nguin | I'd have to go to the manual to find out.` |
| 21:25.06 | autofsckk | vmwi an mwi serv on supplementary righ? |
| 21:25.27 | autofsckk | what does mwi means? |
| 21:26.13 | Qwell | ~mwi |
| 21:26.13 | infobot | Message Waiting Indicator |
| 21:26.15 | autofsckk | turned mwi serv and it took that tone |
| 21:26.21 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 21:26.28 | autofsckk | thanks a lot again :D |
| 21:27.52 | F2Knight | Q: Wondering how to stop SIP phones from ringing when someone is doing a specfic kind of SIP attack... it seems they are scanning the network and sending SIP INVITES to all the items it finds.. this is causing the handsets to ring ... (these handsets are remote connecting devices mind you) Anyone have an idea how to prevent this ? |
| 21:28.45 | r0m|u | alwaysauthreject=yes |
| 21:29.00 | r0m|u | allowguest=no |
| 21:29.12 | r0m|u | F2Knight, ^^ |
| 21:29.23 | r0m|u | I am sure somebody will correct me if I am wrong |
| 21:30.00 | r0m|u | ooo well in that case the phones them self are been scanned? |
| 21:30.06 | r0m|u | not asterisk? |
| 21:30.10 | r0m|u | F2Knight, ^^ |
| 21:30.22 | autofsckk | p3nguin: last thing, when i place a phone call, the call takes maybe 5 or 8 secs to start ringing, i think that should be modified in the spa right? |
| 21:31.39 | autofsckk | PSTN Ring Timeout: maybe this? |
| 21:32.07 | autofsckk | or thes one? PSTN Ring Thru CWT Delay: |
| 21:32.45 | p3nguin | First of all, there is NO REASON that SIP phone should be reachable from outside. |
| 21:33.39 | [TK]D-Fender | lcheeeckout time, BBIAB |
| 21:33.48 | p3nguin | autofsckk: Sounds like a dial plan issue to me. You remember how I told you about S0 and S1 and S2 before? Use it as I instructed you to use it. |
| 21:35.46 | autofsckk | p3nguin: yes i remember that, i putted S0 and it is still slow |
| 21:36.22 | autofsckk | i moved the value at PSTN Ring Timeout: at seems to fix a little |
| 21:38.14 | autofsckk | p3nguin: as a matter of fact, it sends the call right away, but now asterisk is the one that takes some time bridging the 2 peers |
| 21:39.17 | r0m|u | :/ si no es juan es pedro. :) |
| 21:43.15 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-cbtmxaevqifonjpn) |
| 21:46.56 | *** join/#asterisk Jupe (~rez@fl-71-55-208-129.dhcp.embarqhsd.net) |
| 21:49.09 | mateu | discovers: sort=alpha; for moh |
| 21:50.05 | p3nguin | There's also random, if you'd prefer that. I know I do. |
| 21:50.41 | Jupe | I'm using the tutorial at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-APT%28Debian%2FUbuntu%29 to install Asterisk on Debian Linux. The command 'add-apt-repository' apparently isn't valid, so I tried to update my manually add the asterisk respositories and do apt-get update. I copied and pasted the repositories into /etc/apt/sources.list |
| 21:50.54 | Jupe | I'm getting error messages like "Err http://packages.asterisk.org `lsb_release/-cs`-proposed i386 Packages 404 Not Found [IP: 76.164.171.229 80]" when I run apt-get update |
| 21:50.54 | leifmadsen | p3nguin: I used alpha the other day when I wanted to play a file, followed by a separate file, then back to the original :) |
| 21:52.55 | Jupe | i copied and pasted the repo's into my sources.list file. for example: deb http://packages.asterisk.org/deb `lsb_release -cs` main |
| 21:53.41 | wcselby | yawns |
| 21:53.56 | leifmadsen | Jupe: what does lsb_release -cs return? |
| 21:54.13 | Jupe | lsb_release: error: No arguments are permitted |
| 21:54.23 | leifmadsen | here you go |
| 21:54.31 | leifmadsen | sounds like an old version then |
| 21:54.39 | leifmadsen | s/here/there/ |
| 21:54.55 | Jupe | an old version of which software/? |
| 21:55.04 | leifmadsen | Ubuntu |
| 21:55.16 | p3nguin | There he went. |
| 21:55.20 | leifmadsen | that's my guess anyways |
| 21:55.29 | leifmadsen | if lsb_release doesn't permit options... |
| 21:55.32 | Jupe | debian 6.0.3 |
| 21:55.46 | tzanger | leifmadsen! |
| 21:55.50 | leifmadsen | tzanger: omghai! |
| 21:56.30 | tzanger | howgoes |
| 21:56.35 | leifmadsen | Jupe: then I suggest changing `lsb_release -cs` to squeeze |
| 21:56.47 | leifmadsen | tzanger: oh it goes -- about to get away from the computer soon and eat dinner, then go to volleyball |
| 21:57.17 | tzanger | sounds like a plan... have a moment for a sip question before you go though? |
| 21:58.17 | Jupe | beach vollyball? or indoor? |
| 21:58.29 | tzanger | beach? canada? surely you jest |
| 21:58.33 | leifmadsen | ya :) |
| 21:58.34 | Jupe | lol |
| 21:58.35 | tzanger | even on our beaches we ice skate |
| 21:58.36 | leifmadsen | it's November! |
| 21:58.45 | Jupe | remember remember |
| 21:58.54 | Jupe | the fifth of november? :D |
| 21:59.01 | leifmadsen | I play beach in the summer, but indoor in the winter :) |
| 21:59.12 | leifmadsen | ok dinner time!!!!! |
| 21:59.17 | leifmadsen | <-- food in ma bellay! |
| 21:59.51 | p3nguin | Potluck time! |
| 21:59.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:59.59 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 22:00.17 | r0m|u | p3nguin, cya. |
| 22:00.38 | r0m|u | I am out guys. cya |
| 22:00.46 | wcselby | later |
| 22:05.54 | Jupe | okay. my debian is not out of date. |
| 22:06.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-wxmdleqsustpoxqo) |
| 22:06.21 | Jupe | so then i should probably change my `lsb_release -cs` to squeeze, as leif said |
| 22:07.25 | Jupe | oh. it does return "Squeeze". i just accidentally added the word "return" to the end of that line. lol |
| 22:07.32 | Jupe | okay. back to the old drawing board. |
| 22:07.51 | Jupe | how the heck do i obtain those dependencies apt-get upgrade can't find? |
| 22:09.39 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:20.28 | *** part/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 22:20.42 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-217-233.inter.net.il) |
| 22:36.53 | F2Knight | r0m|u, I am not looking for something on asterisk.. this is for the clients phones... ware attackers are scanning large blocks of IP addresses, they are getting some phones to 'ring' my theory is it is because they are sending a SIP INVITE attacek as I have replicated this effect with a modified sip vicious tool |
| 22:37.04 | navaismo | the HPEC its like g729 codec you need to buy 4 license for use with a 4 port card? |
| 22:38.44 | F2Knight | p3nguin, if SIP were not reachable from the outside, you would have no VoIP. you HAVE to allow sip connections, SIP trunks , external extensions etc.. but that means an external phone will have a hole in the firewall of there router.. depending on the router... it may allow it to be reached. |
| 22:40.29 | wcselby | navaismo I'm pretty sure you can buy jsut one license to use with a 4 or 8 port card |
| 22:40.39 | navaismo | excelent |
| 22:40.43 | wcselby | you're licensing the technology, not the channel |
| 22:40.44 | navaismo | ty wcselby |
| 22:41.09 | wcselby | F2Knight yeah that's the way it is |
| 22:41.25 | wcselby | but there are ways to have voip without open holes in your firewall |
| 22:41.45 | wcselby | you can use a more restrictive policy to only open holes for specific IP addresses / ranges, etc. |
| 22:41.55 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 22:41.57 | wcselby | you can use vpn tunnels to encrpyt the traffic, etc |
| 22:43.21 | F2Knight | wcselby, the problem at this point is not the asterisk box. it is the remote clients. |
| 22:43.55 | F2Knight | asterisk (public-ip) --- internet ---- client(router)---IP Phone |
| 22:44.14 | wcselby | F2Knight yeah I know what you mean, I get that too, we actually had a conversation about it in here a few days (weeks) back |
| 22:44.44 | F2Knight | what seems to be happeneing is people are scanning large blocks of IP's with tools like sip vicious using an INVITE packet to see if they get a response.. so they are hitting the CLIENTS ip.. and the phones ring. |
| 22:44.57 | F2Knight | what was the resolve? |
| 22:46.06 | wcselby | put the phones behind a better router that doesn't leave connections over port 5060 open to the world |
| 22:48.47 | JT | F2Knight: just put phones behind a NAT router on private IPs |
| 22:48.54 | JT | then randoms can't get at them |
| 22:51.19 | *** join/#asterisk coppice (~chatzilla@14.63.101.60) |
| 22:52.07 | F2Knight | JT thats apperntly not always true. |
| 22:52.52 | JT | it's generally true unless the devices behind NAT have backdoors into them that make connections out or the NAT router isn't secured |
| 22:52.56 | F2Knight | JT, case in point I have a phoen that this happens to as well, thought they were making it up for the longest, but our asterisk servers were not sending any data, did packet captures for a few days to confirm. its attacking the IP directly |
| 22:52.57 | JT | or you have port forwards |
| 22:53.16 | JT | you must have port forwards |
| 22:53.25 | JT | that doesn't make any sense otherwise |
| 22:53.52 | JT | you can't attack a private IP from the Internet unless it's already opened a connection to you |
| 22:54.19 | F2Knight | jt, I understand the princable .. but I have personally experiensed this.. |
| 22:54.34 | JT | or you forge source addresses that you know have legitimate traffic and even then you won't do much as you won't get replies |
| 22:54.44 | JT | you'll need to be more specific in how it works |
| 22:54.51 | F2Knight | I am behind a router.. no port forwarding no DMZ natted, and it happens about 2 times a week |
| 22:55.18 | JT | so what happens, a phone rings and there's no audio? |
| 22:55.35 | F2Knight | correct. |
| 22:55.49 | F2Knight | it will justs light up all the lines, and ring endlessly |
| 22:56.02 | JT | what does a packet capture show is happening? |
| 22:56.08 | F2Knight | use to say 'unknown' as the callerid.. lately they have changed to 'sip' |
| 22:56.24 | F2Knight | well I can not do packet captures on clients end points. |
| 22:56.37 | JT | no? |
| 22:56.42 | JT | it'd be insightful |
| 22:57.09 | F2Knight | it would I am sure. but I am not walking a 15 dollar a month client through setting up equipement to debug something on there side. |
| 22:57.54 | JT | then you're obviously not worried, don't worry about it :) |
| 22:58.14 | F2Knight | well the calls about it are getting annoying. |
| 22:58.22 | F2Knight | about 20 a day at this point |
| 22:58.26 | JT | one day you'll worry enough |
| 22:58.28 | ruied | F2Knight, why not using phones with openvpn? there are some phones with openvpn implemented.... |
| 22:58.33 | JT | and find out what's happening |
| 22:58.50 | JT | ruied: unnecessary complexity for most situations i think |
| 22:59.01 | F2Knight | ruied, not pratical |
| 22:59.25 | JT | F2Knight: anyway, we can't come up with a reasonable solution without knowing what the problem is |
| 22:59.32 | JT | all we know is what the symptoms are |
| 23:00.14 | F2Knight | well like i said I have verified that if i use sip viscious and attack the phone I can make it ring |
| 23:00.15 | ruied | to me it seems very pratical, and no nat problems.... |
| 23:00.29 | _Corey_ | F2Knight: Are your phones directly on the Internet? |
| 23:00.40 | JT | how does it attack the phone? |
| 23:00.41 | F2Knight | ruied, what the cost to replace 300+ phones? |
| 23:00.58 | JT | ruied: nat works fine most of the time these days at least :) |
| 23:01.04 | F2Knight | _Corey_, some are but most are behind a nat right on the clients lan |
| 23:01.32 | ruied | F2Knight, how many sytes do you have working with that phones? |
| 23:01.33 | F2Knight | business clients with a PBX have no problems... its just the onces that connect remotelly to the hosted services... |
| 23:01.45 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:01.51 | _Corey_ | F2Knight: Why don't you just use a firewall to only allow the remote PBX(s) to talk to the phone? |
| 23:01.52 | F2Knight | ruied, with that manufacture of phone.. about 100, |
| 23:02.45 | F2Knight | _Corey_, because it is the clients equipement we have no authorization or access to modify them |
| 23:03.24 | JT | F2Knight: surely the only way this could work is if the client has a defective NAT router |
| 23:03.31 | JT | but perhaps i'm misunderstanding the problem |
| 23:03.33 | F2Knight | your suggestion would be like having comcast tell you to give them acces to modify your router to block traffic... not a very likely thing to let happen |
| 23:04.03 | F2Knight | JT, I would have presumed soo my self. and for the better part of year I have brushed it off as such |
| 23:04.23 | F2Knight | but I started gettnig them about a month ago, and with incresing frequency |
| 23:05.28 | JT | F2Knight: i'm really curious how it actually works |
| 23:05.52 | F2Knight | You and me both! |
| 23:08.02 | _Corey_ | F2Knight: Well, config your phones to use something other than 5060 for SIP |
| 23:08.19 | _Corey_ | F2Knight: The PBX won't care when they register... the scanner won't find them |
| 23:08.21 | JT | F2Knight: i think it involves packet capture :P |
| 23:08.41 | JT | or installing Snoms and a way to get to Snoms remotely |
| 23:08.47 | F2Knight | _Corey_, again not a pratical option as the sip phones are not always able to be remotely configured, |
| 23:08.48 | JT | snoms have inbuilt tcpdump support |
| 23:09.17 | F2Knight | phone replacement is not an option. |
| 23:09.34 | JT | you're quite a no man aren't you? :P |
| 23:09.37 | JT | everything is can't be done |
| 23:09.42 | JT | i meant just for 1 user |
| 23:09.46 | _Corey_ | F2Knight: Well, the phones are getting the calls directly... if you can't even modify the phone configs to prevent them from taking the calls and you can't firewall the phones it sounds to me that you're SOL |
| 23:09.47 | JT | who was getting attacked |
| 23:09.57 | wcselby | sorry for dropping out of the convo |
| 23:09.59 | wcselby | the issue is this |
| 23:10.00 | F2Knight | for 1 user not an issue, but for several 100 it is not. |
| 23:10.16 | wcselby | when the phone registers to the asterisk server, it's creates an open tunnel through the nat router on port 5060 or whatever |
| 23:10.17 | JT | we just need to find out what is going on |
| 23:10.19 | JT | for one user |
| 23:10.20 | JT | any user |
| 23:10.29 | wcselby | some really poorly designed nat routers will leave that connection open and available to the world to scan |
| 23:10.47 | wcselby | and when the nat router receives another request on port 5060, will forward it back to the phone |
| 23:10.49 | F2Knight | _Corey_, not all the phones are accessable to remotely configure... some pull a config from us others were statcikly set ages ago. some are softphones etc. |
| 23:10.56 | JT | wcselby: that's what i'm thinking |
| 23:11.03 | wcselby | that's exactly what's happening |
| 23:11.14 | JT | wcselby: a good nat router should associate a nat translation with only one foreign external IP |
| 23:11.14 | wcselby | and there's nothing that can be done about it, other than replace the router with something that doesn't do that |
| 23:11.19 | JT | yep |
| 23:11.21 | wcselby | however |
| 23:11.26 | JT | defective NAT router :) |
| 23:11.43 | wcselby | when general residential ISP, such as AT&T, ship out routers that have this issue, there' snothing that the end user can do about it |
| 23:11.53 | wcselby | i know, I have at&t at home, and have this issue often |
| 23:12.11 | wcselby | i think some residential and business comcast routers do the same thing |
| 23:12.17 | F2Knight | JTi would go with the defective nat routers, except I am using a SG580, and have had it happen as well.. |
| 23:12.41 | JT | F2Knight: work out how it happens with that then :) |
| 23:13.22 | wcselby | i just explained how it happens, and there's not much that can be done about it. |
| 23:13.43 | wcselby | the other thing is this - it doesn't seem to cause any problems, the scanner is looking for a way to make outbound calls, they can't do that by just sending an invite |
| 23:13.43 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 23:13.45 | F2Knight | I suppose I could put the phone on its own port and just dump that port... have to see if the route supports it or not. |
| 23:13.54 | wcselby | i think it's more ofa a side effect of the type of scan the attackers are doing |
| 23:13.58 | wcselby | than a goal |
| 23:14.08 | F2Knight | no it doesnt cause a toll problem. |
| 23:14.23 | F2Knight | it causes a problem for people getting random blasts of calls at 3 or 4 am |
| 23:14.29 | JT | it causes a support problem |
| 23:14.31 | wcselby | F2Knight yeah I agree |
| 23:14.31 | F2Knight | waking them up and pissing htem off |
| 23:14.38 | wcselby | and them in turn calling you |
| 23:14.50 | wcselby | i get that |
| 23:14.57 | F2Knight | agreeed its a side effect. easier to scan everything then to look for a specfic thing |
| 23:15.19 | F2Knight | wcselby, exactly. |
| 23:15.27 | wcselby | the best solution would be to put a proxy out on the site, an option most customers aren't willing to pay for |
| 23:15.32 | JT | F2Knight: are the INVITEs matching the extension number the user has? |
| 23:15.33 | wcselby | and have the proxy disregard the attacks |
| 23:15.44 | wcselby | JT nope |
| 23:15.45 | JT | proxy is overkill, an ACL will thwart it. |
| 23:16.04 | wcselby | well, okay, what I guess I should have said was better equipment :) |
| 23:16.05 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:16.10 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 23:16.13 | wcselby | be that a proxy, or a better router / firewall,e tc |
| 23:16.24 | _Corey_ | F2Knight: Well, you're not going to get the scanners to stop... we've seen them increase in frequency and had just what you've described happen to a university client with an open network. No solution but to change the port on the phone or put something (firewall etc) between the phones and the Internet |
| 23:16.36 | wcselby | the invites are just random scans |
| 23:16.39 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 23:16.50 | wcselby | if the phone itself can be told to only allow sip connections from one ip, that may be an option |
| 23:17.29 | F2Knight | wcselby, that is kind what I was thinking,,, a way to tell the phone to only accept invites from the 'account' it is registered too |
| 23:17.37 | wcselby | _Corey_ yeah, that's what I usually do as well. |
| 23:17.56 | _Corey_ | Well, if you can't configure the phones it's pretty much a catch-22 |
| 23:17.56 | wcselby | F2Knight if you figure out how to do this with a polycom 550, please share :) |
| 23:18.06 | F2Knight | that wouldnt be a problem for most situations. where you manage the network, but this is not that case. |
| 23:18.46 | F2Knight | we have lots of the Grandstream GXP seriies phones in deployment.. hoping to get an answer back soon from them. (opened ticket before coming in here) |
| 23:19.09 | JT | oh gawd |
| 23:19.16 | JT | grandstream, people still use them? :P |
| 23:19.26 | Naikrovek | indeed they do |
| 23:19.30 | JT | ~gs |
| 23:19.31 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:20.01 | wcselby | that infobot line is years old, and disregards many improvements to both hardware and firmware |
| 23:20.14 | wcselby | many *recent* improvements, meant ot say |
| 23:20.20 | JT | i still can't think of any good reasons to use them |
| 23:20.47 | ruied | F2Knight, with the same price you can have yealink ones with VPN... |
| 23:20.57 | _Corey_ | If you want a desktop video phone for under $500 your options narrow |
| 23:21.13 | JT | the yealink ones are quite decent |
| 23:21.17 | JT | TI DSP chipset |
| 23:21.43 | coppice | Grandstream also use TI, so what relevance does that have? |
| 23:22.02 | JT | the same chipset? |
| 23:22.05 | F2Knight | grandstream offeres the best price / feature set. they have a good 'feel' to them and are affordable for an office to grow with. 160 for a GS 360 for a snome.. the cost point means a client will deploy more of them, enabling more people to work. and they are often better then the linksys crap phones providers try to 'give' |
| 23:22.34 | JT | F2Knight: you can get polycoms for less |
| 23:22.49 | JT | than $160 |
| 23:22.52 | wcselby | and you know what? |
| 23:22.57 | wcselby | macs are better than pc's |
| 23:23.01 | wcselby | and linux is better than windows |
| 23:23.02 | F2Knight | jt cant stand polycoms / cisco |
| 23:23.09 | JT | F2Knight: why's that? |
| 23:23.11 | wcselby | and I shot the sheriff, but not the deputy |
| 23:23.20 | F2Knight | hate the config process |
| 23:23.25 | JT | wcselby: that's not the same at all? |
| 23:23.34 | ruied | JT, I have about 100 of gs phones, no problems wit one single unit. They are not goog looking and the any alteration at the phone/ata requires a restart, aside that, I can live with them... |
| 23:23.34 | F2Knight | almost no feature sets in the phones |
| 23:23.36 | _Corey_ | I'd definitely dispute that they have the "best" feature set & price, but I do have to agree GS has improved considerably over the last few years |
| 23:23.37 | JT | F2Knight: they seem to mass provision just fine? |
| 23:23.44 | wcselby | sure it is, at this point it's a matter of which do you personally prefer |
| 23:23.53 | wcselby | or more importantly, what does the client prefer |
| 23:24.28 | F2Knight | jt .. mass provision yes. but configure just 2 or 4 at a time. |
| 23:24.35 | wcselby | i personally prefer working with polycoms, and will recommend them every time |
| 23:24.47 | wcselby | i know people that prefer working with ciscos, and will recommend them every time |
| 23:24.54 | wcselby | same with grandstream, snom, and yealink |
| 23:25.00 | F2Knight | wcselby, i think i have 2 301's in a box some place.. |
| 23:25.01 | JT | F2Knight: you're disregarding them because you dont like the web interface? |
| 23:25.05 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-kendzvoxvdbnnvkc) |
| 23:25.09 | wcselby | as long as they function at a level that the end user is happy with, what does it matter? |
| 23:25.16 | JT | i don't mind polycom, cisco, snom, aastra |
| 23:25.21 | JT | they all have their benefits |
| 23:25.26 | JT | even linksys can be ok ;) |
| 23:25.33 | wcselby | linksys is now cisco, btw |
| 23:25.37 | wcselby | :P |
| 23:25.41 | JT | different phone serious |
| 23:25.41 | coppice | If someone brings out a really great IP phone it will be the first |
| 23:25.46 | F2Knight | jt/ no the cisco's do not have a web interface.. the polycoms... seem to have a weak interface. |
| 23:25.47 | JT | they're still a linksys underneath |
| 23:25.49 | wcselby | it's still a cisco 50x |
| 23:25.51 | wcselby | not a linksys 50x |
| 23:26.06 | F2Knight | yea cisco needed a cheap consumer unit |
| 23:26.07 | JT | F2Knight: i've got a cisco here on my desk, definitely has a web interface |
| 23:26.12 | JT | not a config interface |
| 23:26.12 | Naikrovek | F2Knight: the latest version fo the polycom firmware makes the polycom web interface MUCH better |
| 23:26.19 | JT | but can read statistics and logs |
| 23:26.21 | F2Knight | JT 79xx? |
| 23:26.27 | JT | 6901 |
| 23:26.27 | F2Knight | 7940/7960? |
| 23:26.36 | JT | you need to enable the interface in provisioning |
| 23:26.37 | wcselby | the 79xx worked well too if you got the config right |
| 23:26.49 | F2Knight | maybe the newer ones do the old ones did not tftp only |
| 23:26.53 | wcselby | i've worked with lots of 7940, 7960, 7941 and 7961 |
| 23:26.55 | JT | the 6901 is a great phone for simple mass rollout applications |
| 23:27.02 | JT | less than AUD$50 |
| 23:27.11 | JT | when buying over 100 anyway |
| 23:27.16 | F2Knight | and that config changed every damn firmware upgrade and broke things.. just ugly all around |
| 23:27.18 | wcselby | i could mass rollout all of my cisco phones too (the 79xx's) |
| 23:27.38 | JT | pretty much all phones can mass rollout :P that's not my point |
| 23:27.42 | wcselby | heh |
| 23:27.52 | JT | it's a good phone for accomodation and utility |
| 23:28.00 | JT | like mounted on a wall somewhere |
| 23:28.04 | wcselby | the 6901 is a cisco? |
| 23:28.10 | JT | yep |
| 23:28.16 | JT | lowest end current model |
| 23:28.26 | JT | very solid |
| 23:28.36 | JT | there is NO info on getting it working with asterisk on the web |
| 23:28.40 | wcselby | does that one do sip or only siinny? |
| 23:28.45 | JT | but it wasn't much different to the 79xx series |
| 23:28.46 | wcselby | skinny* |
| 23:28.55 | JT | nah cisco have both sccp and sip fw |
| 23:28.59 | JT | comes with sccp loaded |
| 23:29.20 | JT | the smartnet contract is like $5 |
| 23:29.22 | F2Knight | do you still have to be a registered cisco partner to get the firmwares? |
| 23:29.27 | wcselby | i've heard (no confirmation) that cisco has dropped all non-CallManager sip support on the 79xx series, and now if you want, you're directed to the 5xx series |
| 23:29.38 | JT | you just need to be a cisco customer with a smartnet contract F2Knight |
| 23:29.56 | wcselby | just need a smartnet contract |
| 23:30.00 | JT | wcselby: :o that would be quite mean, then again it is cisco |
| 23:30.02 | wcselby | i think I even link to the contract on my blog |
| 23:30.06 | wcselby | the contract you can buy from cdw |
| 23:30.16 | F2Knight | dont agree with that practice.. if a client buys the phone they should get firmware updates when made available with out having to buy a contract |
| 23:30.41 | wcselby | not all vendors agree with you there |
| 23:30.55 | coppice | there should be laws to prevent this kind of customer abuse |
| 23:31.01 | F2Knight | this is true.. and another reason why I prob. would not use there equipment. |
| 23:31.07 | wcselby | even polycom doesn't release the newest and greatest for free, you have to wait a couple generations unless you're a parter or something |
| 23:31.30 | F2Knight | and polycom makes cisco phones still i think |
| 23:31.43 | wcselby | http://www.cdw.com/shop/products/Cisco-SMARTnet-extended-service-agreement/993513.aspx?enkwrd=ALLPROD%3aCON-SNT-CP7941 here is the smartnet on cdw for a 79x1 phone, I think |
| 23:31.44 | F2Knight | evil vs evil, |
| 23:32.01 | coppice | polycom only makes the cisco conferencing phones |
| 23:32.07 | F2Knight | part of why i refuse to use any of there equipment. |
| 23:32.11 | JT | cisco have their quirks, but evil? o_O |
| 23:32.42 | wcselby | cisco has some jacked up policies about the legalities of using their phones in a non-callmanager environment |
| 23:32.45 | wcselby | ~cisco |
| 23:32.45 | infobot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 23:32.53 | wcselby | oh what |
| 23:32.59 | wcselby | not what I thought it was gonna be |
| 23:33.00 | wcselby | lol |
| 23:33.22 | F2Knight | LOL |
| 23:33.25 | Naikrovek | wcselby: you have to wait one release cycle. when 4.0.1 is out for dealers, 4.0.0 will be out for everyone |
| 23:33.33 | wcselby | Qwell has an open letter on the net somewhere with a response from cisco basically stating they can sue your pants off if you use a 79xx phone not in a call manager environment |
| 23:33.52 | wcselby | Naikrovek yeah that's what I was trying to say |
| 23:33.59 | Naikrovek | they say that you still have to pay the licensing, even if you use it for asterisk |
| 23:34.19 | Naikrovek | so you ahve to pay for the call manager license even when you use a cisco phone for Asterisk. |
| 23:34.21 | F2Knight | wcselby, like infobot said.. to be avoided . |
| 23:34.25 | Naikrovek | yup |
| 23:34.36 | Naikrovek | with 4.0.0, Polycoms are finally all the way there. |
| 23:34.50 | Naikrovek | SIP REASON header supported, can change the config without rebooting the phone (mostly) |
| 23:34.55 | Jupe | infobot tell me about sip |
| 23:34.55 | Naikrovek | lots of other awesome things |
| 23:35.06 | Jupe | hehe. i love infobots. haven't used one in a long time |
| 23:35.14 | wcselby | Naikrovek they had the config without reboot thing going since 3.3 I think |
| 23:35.23 | Naikrovek | yes, but it's expanded even more now |
| 23:35.26 | Naikrovek | only a very few things require a reboot |
| 23:35.28 | wcselby | nice |
| 23:35.39 | Naikrovek | oh you can do hot desking on the phone now, without support from asterisk |
| 23:35.40 | Naikrovek | lots of stuff |
| 23:35.41 | wcselby | i'm pretty sure i'm running 3.3 on my home 550 |
| 23:35.41 | JT | so polycoms are evil.... cos we need to wait 1 release cycle to get the firmware? |
| 23:35.46 | Naikrovek | i recommend reading the release notes. |
| 23:35.59 | wcselby | i didn't say they were evil |
| 23:36.00 | Naikrovek | JT: it's just so that people who can support new versions get it first. |
| 23:36.08 | wcselby | i think someone else said they were evil for making stuff for cisco |
| 23:36.10 | wcselby | or something |
| 23:36.45 | Jupe | infobots can be fun/annoying toys on casual chat channels |
| 23:36.46 | F2Knight | I said they were evil. |
| 23:37.11 | JT | F2Knight: a bit religious with your phones? |
| 23:37.26 | JT | Naikrovek: i don't have a problem with that, seems reasonable |
| 23:37.32 | F2Knight | not so much religious.. more anti regeme |
| 23:37.32 | JT | to iron out the bugs |
| 23:37.37 | Naikrovek | regime* |
| 23:37.51 | JT | well why but anything from a corporation/regieme |
| 23:37.56 | wcselby | no he hates it when geme's repeat |
| 23:37.57 | JT | go be self sufficient ;) |
| 23:38.07 | JT | s/but/buy/ |
| 23:38.13 | wcselby | laughs |
| 23:38.13 | Naikrovek | wcselby: geme repeating is for wankers |
| 23:38.20 | Naikrovek | :P |
| 23:38.26 | wcselby | lol |
| 23:38.35 | wcselby | anyways |
| 23:38.36 | F2Knight | no I think it is evil of a company to sell something to someone then charge them to download the updates. |
| 23:38.38 | wcselby | it's time to roll on out of here |
| 23:38.55 | wcselby | o/ peace out folks |
| 23:39.00 | Naikrovek | F2Knight: they don't charge to download the updates, they use their resellers (who CAN support problems) beta test the software for public release |
| 23:39.11 | F2Knight | in the same way i think it is evil of apple to restrict, monitor and control what content can be placed in the app store for sale |
| 23:39.18 | F2Knight | or what software can run on the device. |
| 23:39.21 | JT | F2Knight: polycom isn't cisco |
| 23:39.24 | JT | the firmware is free |
| 23:39.29 | Naikrovek | ^^^^^^^^ |
| 23:39.30 | beek | thankfully! |
| 23:39.34 | Naikrovek | aye |
| 23:40.07 | *** join/#asterisk dhorner_mb (~dhorner_m@184.18.34.27) |
| 23:40.23 | F2Knight | sense when did they start releasing the firmware as free downloads to the public? |
| 23:40.31 | Naikrovek | support.polycom.com |
| 23:40.35 | Naikrovek | for years and years |
| 23:40.45 | F2Knight | I was talking about cisco |
| 23:40.57 | Naikrovek | cisco doesn't |
| 23:41.01 | Naikrovek | ... i dont' think |
| 23:41.05 | JT | F2Knight: over 5 years |
| 23:41.18 | JT | or since ever? |
| 23:41.41 | JT | so since polycom don't charge for their firmware |
| 23:41.45 | JT | does that mean they're not "evil"? |
| 23:42.22 | F2Knight | that single fact alone would not preclude them from it but it would make them less so based on that basis alone yes. |
| 23:43.33 | F2Knight | I personlly do not like PolyCom web interface on the 301, (granted the only polycom i have used) but it ws easier to get working then cisco. supported sip, and did not have the big brother control and feel of cisco |
| 23:43.57 | JT | yeah |
| 23:44.04 | JT | ciscos problem is they are undocumented |
| 23:44.09 | JT | as they want you to use CUCM |
| 23:44.17 | F2Knight | whom if you wanted to update the firmware on a phone you need to purchase a contract from them, and then have mounds of legal BS to deal with still |
| 23:44.25 | JT | polycoms have excellent documentation and it's officially supported and encouraged on the other hand |
| 23:44.59 | F2Knight | my 301 is nice. but it has no where near the features of the GXP2110 from grandstream. |
| 23:45.18 | JT | 301 was a very basic phone and is over 7 years old now i think |
| 23:45.37 | Naikrovek | yeah |
| 23:45.40 | F2Knight | yea I think it is too.. but its been in my closet for about 4 years |
| 23:45.44 | Naikrovek | 335 or 450 are SOOOO much better |
| 23:45.55 | *** part/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 23:46.04 | F2Knight | so .. |
| 23:46.25 | F2Knight | should a phones SIP stack only accept INVITES from the device it is registerd too? |
| 23:46.33 | F2Knight | or should it accept if from ANY device. |
| 23:46.58 | JT | it should only accept invites that are To: its address |
| 23:47.03 | JT | since it's not a B2BUA |
| 23:47.07 | JT | or proxy, imho |
| 23:47.45 | p3nguin | If it's not To: its address, how else could it get to it? |
| 23:48.04 | F2Knight | I am looking in the GXP configs... i see this Direct IP Cals. |
| 23:48.43 | Naikrovek | oh |
| 23:48.48 | Naikrovek | you're questioning the SIP protocol then |
| 23:48.52 | F2Knight | so I am wondering.... |
| 23:49.05 | JT | p3nguin: @192.168.0.38 for example |
| 23:49.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:49.19 | JT | To: 123@192.168.0.38 |
| 23:49.37 | JT | if the phone doesn't have a 123 line appearance |
| 23:49.40 | JT | it shouldn't answer |
| 23:49.45 | JT | or ring |
| 23:49.46 | JT | etc |
| 23:50.05 | F2Knight | have any of you used sip vicious? |
| 23:50.27 | Naikrovek | no |
| 23:50.32 | F2Knight | http://code.google.com/p/sipvicious/ |
| 23:50.35 | Naikrovek | clever name, though |
| 23:50.47 | F2Knight | give it a wherel on your network, simple python script |
| 23:51.03 | Naikrovek | but to answer your question, I think you can disable the direct sip call thing on polycoms |
| 23:51.07 | Naikrovek | though i dont' know how |
| 23:51.23 | F2Knight | has a map tool will scan entire IP blocks, even give you finger prints.. or jsut modify the script a little and war attack an entire network |
| 23:52.13 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 23:52.35 | Naikrovek | "currently consists of four tools." has 5 tools. |
| 23:52.42 | Naikrovek | attention to detail :) |
| 23:52.55 | Naikrovek | i'll give it a go |
| 23:53.18 | F2Knight | svreport is not really a user tool |
| 23:53.33 | F2Knight | svcrack svcrash svmap and svwar |
| 23:54.12 | JT | Naikrovek: i think the problem is the phone is ringing out loud when INVITES To: any number are received by it |
| 23:55.31 | F2Knight | exactly |
| 23:55.46 | JT | unless they have common extension numbers |
| 23:55.52 | JT | and keep getting hit by the scanner |
| 23:56.16 | F2Knight | nah the 'extension' numbers are there phone numbers. an attacker wouldnt know what that was by just scanning |
| 23:56.52 | JT | yeah and i doubt they'd be scanning that high a number too.. |
| 23:57.01 | F2Knight | example... |
| 23:57.06 | Naikrovek | ah i see, now. okay. well a firewall could cure that. |
| 23:57.30 | F2Knight | ./svwar.py -e100-9999999999999 -m INVITE ipaddress |
| 23:57.40 | F2Knight | causes my device to ring |
| 23:57.50 | JT | lots of times? |
| 23:57.52 | Naikrovek | i don't have a phone handy, but i'll try this tomorrow |
| 23:58.22 | Naikrovek | i'll be able to tell you if polycoms still do this, then. |
