| 00:14.26 | p3nguin | seri: You could use channel redirect, but you have to know ahead of time what context,extension,priority you're going to send the call to. You'd be better off to configure the dialplan appropriately so the call goes to voicemail on its own. Or if your phone has an ignore button that will send to voicemail if configured to do so. |
| 00:15.59 | p3nguin | My phones have a divert button, that if I press it when the phone is ringing, the call is diverted to voicemail. I can also press the End Call button while it is ringing to stop the ringing, but the Dial() timeout still applies before going to voicemail. |
| 00:20.04 | SeRi | p3nguin: I see. I just wanted to do it to calls comming in in the main line which are in the PAP2 |
| 00:22.42 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 00:23.05 | *** part/#asterisk LemensTS (~matthew@70.238.163.254) |
| 00:23.57 | SeRi | Qwell: ping |
| 00:24.10 | SeRi | any developers in? |
| 00:24.38 | *** join/#asterisk francisvgarcia (~networker@186.1.68.198) |
| 00:25.44 | francisvgarcia | Hi folks |
| 00:26.10 | francisvgarcia | I got a lil issue using the MP3Player for web streaming |
| 00:28.13 | francisvgarcia | I'm having this error |
| 00:28.16 | francisvgarcia | app_mp3.c:133 timed_read: Poll timed out/errored out with 0 |
| 00:28.31 | francisvgarcia | I have mpg123 already installed |
| 00:30.08 | *** join/#asterisk srd (hbunting@ec2-50-18-185-63.us-west-1.compute.amazonaws.com) |
| 00:31.18 | srd | How could I make a conference call by just dialing an extension and have the dialplan call two numbers simultaneously and bridge them? |
| 00:35.00 | SeRi | [TK]D-Fender: what does res_http_post belong to? and why does asterisk-gui need it? |
| 00:35.35 | lanning | srd: probably an AGI script that writes call files, then connects the current channel to the conference |
| 00:35.40 | [TK]D-Fender | *-GUI runs on AJAX |
| 00:36.15 | SeRi | [TK]D-Fender: if I build it and res_http_post is not present it fails |
| 00:37.21 | SeRi | I am trying to understand the relationship between both... |
| 00:38.53 | SeRi | any body knows? |
| 01:01.32 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 01:03.20 | p3nguin | seri: So today I did a tcpdump on the vyatta dhcp server. I clearly see a static route and a default gateway being pushed out on client leases. |
| 01:03.42 | *** join/#asterisk bloudermilk (~bloudermi@pool-108-38-59-34.lsanca.fios.verizon.net) |
| 01:04.07 | p3nguin | That leads me to believe that the dhcp clients must be broken. If not that, then I don't really have an explanation. |
| 01:04.32 | p3nguin | boulder milk? |
| 01:04.38 | bloudermilk | p3nguin: ? |
| 01:04.42 | p3nguin | oh, nevermind. |
| 01:05.00 | bloudermilk | Is this the same p3nguin that frequents the jb scene? |
| 01:05.17 | p3nguin | Probably not, since I don't know what jb is. |
| 01:05.20 | SeRi | p3nguin: I have massive comcast issues |
| 01:05.29 | bloudermilk | fair enough :) |
| 01:05.51 | p3nguin | Though I am curious, now that you mention it. |
| 01:06.55 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 01:07.31 | p3nguin | Would you mind cluing me in? |
| 01:11.54 | *** join/#asterisk em_pleh (~em_pleh@rrcs-67-52-227-134.west.biz.rr.com) |
| 01:11.57 | em_pleh | hello |
| 01:13.25 | em_pleh | I am trying to connect to my pbx from outside the network using IAX2. I have already done port forwarding for "4569" on the firewall and still cannot connect. I have tried from multiple outside computer networks and have had no luck. |
| 01:14.56 | WIMPy | em_pleh: Did you forward 4569 UDP? |
| 01:15.19 | p3nguin | And what are you using to "connect" to asterisk? |
| 01:15.59 | em_pleh | WIMPy yes I did |
| 01:16.05 | em_pleh | p3nguin im using Zoiper |
| 01:16.20 | em_pleh | when i try it from in the network it works just fine |
| 01:16.38 | p3nguin | Sounds like a firewall issue. |
| 01:16.53 | em_pleh | well everything else forwarded on the firewall works like a charm |
| 01:17.22 | WIMPy | Does Asterisk have a way back out? |
| 01:17.36 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 01:17.42 | SeRi | jesus. |
| 01:17.47 | SeRi | this is just wrong |
| 01:18.08 | SeRi | There should be a law against companys liek comcast |
| 01:18.32 | em_pleh | SeRi is it comcast blocking the port? |
| 01:18.46 | SeRi | em_pleh: no. just having serice issues |
| 01:18.47 | em_pleh | WIMPy im not sure i understand the question |
| 01:19.01 | WIMPy | SeRi: They will continue as long as they have enough customers. |
| 01:19.28 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 01:19.32 | WIMPy | em_pleh: Does Asterisk have a route back and is it allowed to reply? |
| 01:19.46 | em_pleh | yes it is |
| 01:19.57 | em_pleh | unless its somewhere in the config i dont know about |
| 01:20.05 | em_pleh | i have even tried changing the port |
| 01:20.12 | SeRi | comcast should disapear from the world |
| 01:20.27 | em_pleh | is there anywhere that I have to allow out of network connections? |
| 01:20.27 | WIMPy | em_pleh: Time to use tcpdump |
| 01:20.30 | SeRi | hide on a sea crab hole or soemthing |
| 01:21.04 | em_pleh | WIMPy shoot away im ready to do |
| 01:21.30 | Micc | anyone familiar with adtran total access? |
| 01:21.40 | p3nguin | em_pleh: firewall |
| 01:21.51 | p3nguin | Make sure the port is not getting blocked. |
| 01:22.08 | p3nguin | Check iptables on the asterisk computer. |
| 01:22.12 | p3nguin | iptables -L -nv |
| 01:22.27 | em_pleh | k let me check |
| 01:22.58 | em_pleh | what am i looking for |
| 01:23.27 | p3nguin | Do you have any rules at all on that system? If so, do you really need them on that system? |
| 01:23.39 | em_pleh | no i dont need any rules |
| 01:23.47 | em_pleh | how do i disable iptables |
| 01:24.02 | em_pleh | WIMPy i used tcpdump on the port and its not even hitting anything on the server |
| 01:24.32 | WIMPy | Well, no request, no reply. |
| 01:24.48 | em_pleh | k must be something in iptables |
| 01:24.54 | em_pleh | p3nguin how do i disable iptables |
| 01:25.02 | p3nguin | Which distro? |
| 01:26.25 | em_pleh | ubuntu |
| 01:26.28 | em_pleh | actually let me check |
| 01:26.43 | em_pleh | I just ran /etc/init.d/iptables stop and it stopped |
| 01:26.47 | p3nguin | Good. |
| 01:27.04 | em_pleh | but still nothing |
| 01:27.09 | em_pleh | IAX2 is udp correct? |
| 01:27.13 | p3nguin | yes |
| 01:27.21 | em_pleh | so how will tcpdump see it? |
| 01:27.55 | p3nguin | Because tcpdump doesn't just see tcp. |
| 01:28.00 | em_pleh | got it |
| 01:28.08 | p3nguin | tcpdump -vv -n port 4569 |
| 01:28.24 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 01:30.12 | em_pleh | ok weird |
| 01:30.20 | em_pleh | i can ping the port and tcpdump sees it |
| 01:30.27 | p3nguin | You can't ping a port. |
| 01:30.39 | em_pleh | nmap -sU --data-length 999 -p2353 mdcphoenix.selfip.com |
| 01:30.45 | em_pleh | darn |
| 01:30.50 | em_pleh | should of cut off the domain lol |
| 01:30.54 | p3nguin | should have |
| 01:31.06 | em_pleh | well thats the ping |
| 01:31.10 | WIMPy | Frre calls! |
| 01:31.27 | p3nguin | It doesn't work, though. :/ |
| 01:31.45 | em_pleh | p3nguin what you mean |
| 01:31.51 | gandhijee | anyone know if there is a polycom phone cfg creator software?? |
| 01:32.01 | p3nguin | You're here because the thing doesn't work. |
| 01:32.09 | em_pleh | right |
| 01:32.22 | em_pleh | but it just shows the firewall lets the port go through |
| 01:32.55 | p3nguin | If you use tcpdump on the asterisk machine, and tell zoiper to register, do you see the packets coming in? |
| 01:33.12 | em_pleh | no not when its from outside |
| 01:33.17 | em_pleh | internal network yes |
| 01:33.38 | p3nguin | You still have a firewall issue, then. |
| 01:33.48 | em_pleh | hummm |
| 01:33.55 | em_pleh | even dough i can send a packet huh |
| 01:34.02 | em_pleh | well im using smoothwall |
| 01:34.07 | bloudermilk | p3nguin: Sorry, was knee deep in logs... JB = iOS jailbreak |
| 01:34.19 | bloudermilk | Though now that I think of it, the guy I was thinking of goes by evil penguin |
| 01:34.47 | bloudermilk | What's the easiest way to watch the SIP traffic for a call? |
| 01:34.48 | p3nguin | I've seen some other people using my nick on forums. |
| 01:35.24 | em_pleh | p3nguin i forward other ports and they work just fine |
| 01:35.32 | em_pleh | only thing not working is this IAX2 port |
| 01:36.16 | *** join/#asterisk fafaflofly (~bababooey@cpe-74-74-198-0.rochester.res.rr.com) |
| 01:36.17 | bloudermilk | ah hah, nevermind :) |
| 01:36.30 | p3nguin | If you watch tcpdump on port 4569, and use nmap to port 4569 from the same computer where zoiper is, do you see the packet? |
| 01:36.41 | em_pleh | yes |
| 01:37.31 | p3nguin | Then zoiper is the part not working? Check the host name and port number configured in zoiper. |
| 01:37.42 | em_pleh | did already |
| 01:37.46 | em_pleh | correct port |
| 01:37.49 | em_pleh | and ip |
| 01:38.11 | p3nguin | Can you get another phone to test? |
| 01:38.19 | em_pleh | I have tryed |
| 01:38.25 | em_pleh | they all work within the network |
| 01:38.27 | p3nguin | I don't know too many iax2 phones, though. |
| 01:38.30 | em_pleh | but not from outside |
| 01:38.55 | em_pleh | now im just wondering if its a permit/deny not allowing localnet to connect |
| 01:39.51 | p3nguin | I don't see how it is even possible for one application's packet (nmap) to reach the asterisk system, but another application on the same computer (zoiper) not reaching it. |
| 01:39.57 | p3nguin | Doesn't even make sense. |
| 01:40.18 | em_pleh | yea and i tried from 2 computers and completly different external networks |
| 01:40.56 | WIMPy | What system are you running zoiper on? |
| 01:41.08 | em_pleh | windows |
| 01:41.17 | WIMPy | ha |
| 01:41.27 | WIMPy | personal firewall? |
| 01:41.32 | em_pleh | nope |
| 01:41.36 | p3nguin | app blocking |
| 01:41.44 | p3nguin | Defender, perhaps. |
| 01:41.54 | em_pleh | firewall is disabled |
| 01:42.01 | em_pleh | and i dont have a 3rd party app |
| 01:44.26 | bloudermilk | Is there such thing as virtualized PSTN? |
| 01:44.32 | p3nguin | Yes. |
| 01:44.34 | p3nguin | ~itsp |
| 01:44.34 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:45.09 | bloudermilk | I meant specifically for virtualizing SS7, etc. |
| 01:45.20 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 01:45.32 | bloudermilk | If I wanted to interface at that level, but not have to get hardware |
| 01:46.44 | SeRi | am I online? |
| 01:46.54 | p3nguin | for now, yes. |
| 01:47.11 | SeRi | hehe... |
| 01:47.17 | WIMPy | SeRi: ping yourself |
| 01:47.24 | SeRi | I am so glad I invested on a GSM gateway |
| 01:47.27 | p3nguin | /ping seri |
| 01:47.33 | SeRi | WIMPy: nice try :P |
| 01:47.44 | p3nguin | Seriously, ping yourself. |
| 01:47.51 | SeRi | as in irc? |
| 01:47.53 | WIMPy | bloudermilk: Such things do exist, but not sure if they are known at that level. |
| 01:47.54 | p3nguin | /ping seri |
| 01:48.06 | SeRi | I thought you guys where talking about as in localhost |
| 01:48.16 | SeRi | lol sorry rough day |
| 01:48.28 | carrar | localhost is a great itsp |
| 01:48.35 | SeRi | hahaha! |
| 01:48.36 | carrar | they are always fast |
| 01:48.52 | SeRi | and 99.98% uptime |
| 01:48.55 | carrar | 100% |
| 01:49.16 | SeRi | no need for localhost reboots? patches? updates? must be nice~ |
| 01:49.22 | SeRi | ! |
| 01:49.39 | p3nguin | I try not to reboot. |
| 01:49.46 | p3nguin | But sometimes I just do it. |
| 01:50.03 | SeRi | I do it when develpoing and testing... bad cod tend to kill systems :P |
| 01:50.16 | carrar | use tuna |
| 01:50.38 | p3nguin | haha |
| 01:50.38 | p3nguin | cod |
| 01:50.48 | SeRi | code* |
| 01:50.58 | p3nguin | loves hand-breaded deep-fried cod nuggets |
| 01:51.17 | Maliuta | I like Barramundi |
| 01:51.30 | carrar | sueshe |
| 01:53.00 | SeRi | 3 hrs in my house and they could not resolve the issue |
| 01:53.06 | SeRi | way to go comcast |
| 01:53.31 | carrar | thats comcast for ya |
| 01:53.47 | SeRi | I been off line most of the day |
| 01:54.36 | carrar | get Frontier Fios 25/25 |
| 01:56.19 | SeRi | I tried to get Fios. way out of my league. |
| 01:56.38 | SeRi | looking at local hosted dsl. |
| 01:57.08 | carrar | Local hosted DSL you can also probably get bonded DSL |
| 01:57.16 | carrar | for twice or triple the speed |
| 01:57.26 | carrar | and better service |
| 01:57.30 | SeRi | indeed |
| 01:57.47 | carrar | anythign is better then comcast |
| 01:57.50 | SeRi | I have a few company's in line. 2012 comes in ill call comast CS and say fuck off |
| 02:15.36 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 02:18.23 | *** join/#asterisk nafg (~quassel@ool-4355e4a2.dyn.optonline.net) |
| 02:18.31 | nafg | Hi, can someone help me diagnose this? |
| 02:18.39 | nafg | channel originate SIP/callwithus/17325342893 application Agi chavrusa.agi |
| 02:18.44 | nafg | [Dec 14 21:12:34] NOTICE[12308]: channel.c:5196 __ast_request_and_dial: Unable to request channel SIP/callwithus/17325342893 |
| 02:19.27 | *** join/#asterisk psharmor (~quassel@97-118-232-206.hlrn.qwest.net) |
| 02:22.32 | ChannelZ | if you did a Dial(SIP/callwithus/17325342893) in your dialplan does it work? (IE, do you have a SIP peer defined called "callwithus" and is that the right number format?) |
| 02:22.40 | ChannelZ | methinks probably not |
| 02:23.27 | *** join/#asterisk master_of_master (~master_of@p57B53FD9.dip.t-dialin.net) |
| 02:33.45 | nafg | ChannelZ: I got it working. I restarted asterisk with an internet connection (last time it booted there wasn't). Thanks! |
| 02:41.33 | *** join/#asterisk coppice (~coppice@m121-202-23-210.smartone.com) |
| 02:45.22 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 02:52.11 | *** part/#asterisk tekoholic (~quassel@97-118-232-206.hlrn.qwest.net) |
| 02:54.39 | *** join/#asterisk nuit123 (~nuit123@ip72-211-223-25.oc.oc.cox.net) |
| 02:55.40 | nuit123 | Anyone out there using DIDforsale.com SIP Trunks? I'm trying to configure a Trixbox for a trunk, but no having much success. |
| 02:56.03 | nuit123 | BTW, my problem probably has more to do with my inexperience not necessarily DIDforsale! |
| 02:56.45 | WIMPy | ~trixbox |
| 02:56.46 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 02:58.19 | nuit123 | is PBXinaFlash or Freeswitch supported here? |
| 02:58.31 | nuit123 | i could wipe and reload my machine. |
| 02:58.36 | WIMPy | no |
| 02:59.05 | WIMPy | The GUIs have their own channels. |
| 02:59.10 | nuit123 | thx |
| 02:59.18 | *** part/#asterisk nuit123 (~nuit123@ip72-211-223-25.oc.oc.cox.net) |
| 03:00.51 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 03:17.14 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 03:26.37 | *** join/#asterisk kayfox (~kayfox@xheotris.zerda.net) |
| 03:28.04 | nafg | Anyone here familiar with AMI? Originating a call from the console works, but from asteriskjava doesn't. |
| 03:28.13 | nafg | Using Live API. |
| 03:28.19 | nafg | No output from callback. |
| 03:30.06 | WIMPy | Try it by hand. |
| 03:31.55 | *** join/#asterisk radic (~radic@dslb-178-002-234-050.pools.arcor-ip.net) |
| 03:32.52 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 03:43.44 | *** join/#asterisk celord (~cesar@celord.ice.co.cr) |
| 03:53.57 | *** join/#asterisk nighty^ (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 03:56.46 | p3nguin | seri: Well, I'm back. Got a shiny new Arch desktop. |
| 03:56.58 | SeRi | p3nguin: !!!!!!!!!! nice! |
| 03:57.05 | p3nguin | 3.1.5-1-ARCH #1 SMP PREEMPT Sat Dec 10 14:43:09 CET 2011 x86_64 |
| 03:57.08 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:57.15 | SeRi | nice |
| 03:57.29 | p3nguin | It wasn't toooo painful. |
| 03:57.43 | SeRi | I am sure |
| 03:57.45 | WIMPy | Now try to compile dahdi on that :-) |
| 03:57.59 | p3nguin | It's a desktop. I have no reason to do so. |
| 03:58.23 | p3nguin | seri: I thought it was totally jacked for a minute. I had to manually remove initscripts before I could get sysupgrade to go, which then had no reason to upgrade initscripts. |
| 03:58.28 | WIMPy | excuses |
| 03:58.37 | p3nguin | So when I started up the new system, it didn't work very well. |
| 03:59.04 | p3nguin | I remembered that I did that, installed initscripts again, and things magically worked again. |
| 03:59.12 | p3nguin | was a little worried for a minute. |
| 03:59.42 | SeRi | ah! cool. |
| 04:00.19 | p3nguin | Anyone know cqkenvox? |
| 04:00.49 | p3nguin | I've been without skype for a week while my shit was busted, and now I have messages from weird people on there. |
| 04:01.00 | nafg | Anyone here familiar with AMI? Originating a call from the console works, but from asteriskjava doesn't. |
| 04:01.00 | SeRi | :/ |
| 04:01.17 | p3nguin | This cqkenvox wants me to allow him to see me when I am online. Stalker? |
| 04:01.18 | nafg | Using Live API, no output from callback. |
| 04:01.30 | SeRi | dijib? |
| 04:01.34 | SeRi | Joking! |
| 04:01.42 | SeRi | lol :P |
| 04:02.42 | p3nguin | So I checked how often I do a sysupgrade... |
| 04:03.04 | SeRi | every 2yrs? |
| 04:03.06 | SeRi | :P |
| 04:03.42 | WIMPy | nafg: Try it by hand. |
| 04:04.07 | p3nguin | [2009-04-28 01:08] starting full system upgrade |
| 04:04.07 | p3nguin | [2010-09-29 12:40] starting full system upgrade |
| 04:04.07 | p3nguin | [2011-12-14 13:52] starting full system upgrade |
| 04:04.41 | SeRi | hahahaha |
| 04:04.42 | SeRi | nice |
| 04:04.57 | p3nguin | Not very often. |
| 04:07.06 | *** part/#asterisk b-scrillz (~lolwut@CPE00226b5cbf94-CM000f9facaeec.cpe.net.cable.rogers.com) |
| 04:07.41 | SeRi | :) |
| 04:08.20 | p3nguin | I still have some polishing to do. I keep finding nicks and scratches on this new system. |
| 04:09.50 | nafg | WIMPy: Ahh, permission denied |
| 04:10.28 | nafg | read=call,write=call not enough for originate? |
| 04:11.14 | nafg | The 1.4 pdf say it needs call, all --- I assume that's either one of course, no? |
| 04:11.43 | WIMPy | There is an "originate" permission. That's probably doing what it says. |
| 04:19.02 | WIMPy | @£*~^£@&! Damn auto keypad feature. |
| 04:19.10 | nafg | WIMPy: Where does it list permissions? |
| 04:19.26 | WIMPy | manager.conf |
| 04:25.48 | nafg | WIMPy: I mean, where does it list the permissions you can put into manager.conf? |
| 04:26.13 | nafg | In any case it worked. I had to add originate and system (at least that AMI said). |
| 04:26.18 | nafg | So thanks again! |
| 04:31.28 | WIMPy | In the sample manager.conf. |
| 04:36.39 | nafg | Ah. |
| 04:38.46 | nafg | Another question: If when AMD detects MACHINE, I do WaitForSilence(2000), most of the message still gets cut off. After listening to my (Sprint) voicemail |
| 04:39.13 | nafg | (all defaults, no custom name or greeting), I decided to add repeat=2. Then it works perfectly. |
| 04:39.53 | nafg | The question is, how safe is it to wait for 2 seconds twice. What if it thinks a person is machine --- he won't hear anything. What if |
| 04:40.08 | nafg | another voice mail service only has one silence? It will never go. |
| 04:40.15 | nafg | Are those real concerns? |
| 04:40.33 | WIMPy | It should be impossible to do that reliably. |
| 04:44.03 | nafg | WIMPy: == it's not safe? |
| 04:44.20 | WIMPy | How could it possibly be safe? |
| 04:44.28 | nafg | What should I do, wait for one silence and play it twice? Wait for a longer silence? |
| 04:44.32 | WIMPy | It's always including some guesswork. |
| 04:44.43 | nafg | Okay, by safe I mean reasonable. |
| 04:45.50 | nafg | When it waits for "two instances" of the two-second silence, does one four-second period of silence count? |
| 04:46.24 | WIMPy | I haven't tried. |
| 04:46.45 | nafg | My second concern can be alleviated by specifying a timeout. |
| 04:48.49 | nafg | Is there a way to "explicitly" specify no timeout? For instance 0? What if you there's a trailing comma, e.g. |
| 04:49.02 | nafg | WaitForSilence(2000,1,) |
| 04:49.58 | *** join/#asterisk gravin (~gravin@175.136.225.116) |
| 04:55.06 | nafg | Doing a timeout of seven seconds, hope it works well! |
| 05:02.37 | phix | GANG! |
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| 05:58.16 | ChannelZ | BANG! |
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| 07:02.15 | IsUp | morning all |
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| 07:49.32 | bajou2_202020 | hi my softphones register on opensips and I use asterisk + a2billing on another server(s) for call billing, and call routing/rates/switching, after that calls are sent to the Tier1 carriers for termination, I am using 10 apache servers on 32bit vmware, would I gain significant improvement in performance using 62 bit OS ...the OS is Centos 5.7 and the Asterisk is 1.8 |
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| 08:02.08 | ChannelZ | You only really would by virtue of being able to put more RAM in the machine |
| 08:02.43 | ChannelZ | assuming your resources are currently tight on that system with a possible max of 4GB |
| 08:04.02 | ChannelZ | (and we'll ignore PAE) |
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| 08:12.37 | bajou2_202020 | ChannelZ, I am good on RAM ...for that purpose I dont really need 64 bit, it was my assumption that 64bit does speed up computations, and Asterisk with A2billing is more or less resource intesive in terms of CPU, that is why I asked about 64bit...I might be barking up the wrong tree though |
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| 09:46.12 | schmidts | good morning |
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| 09:57.30 | ixyd_ | hi guys, iam restructuring my dialplan to only use subs instead of macros (i know 1.8 is already doing this all the time)... iam wondering about the following warning: "....application call to GoSub affects flow of control...." can someone tell me if there is a real reason not to use gosub directly? |
| 09:59.22 | field_it | hi. sometimes everything works in<->out. usually in the morning asterisk does not receive calls from outside (caller gets "no such number" tone). and also in the morning often a call is established from inside out it's only ringing "half" a time and then silence but the channel's open and inside can hear the outside caller but not vice-versa. |
| 09:59.58 | field_it | ideas? |
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| 10:07.19 | schmidts | field_it which version do you use? coudl be a deadlock problem |
| 10:09.26 | field_it | schmidts: asterisk v1.8.7.1 |
| 10:09.42 | field_it | schmidts: why do you think it might be a deadlock? |
| 10:10.09 | ixyd_ | field_it: what is your outbound conenction? sip, dahdi...? |
| 10:10.14 | schmidts | field_it just an idea ;) |
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| 10:11.21 | field_it | ixyd_: ah, thx, good question. its sip. |
| 10:11.26 | field_it | schmidts: ok |
| 10:11.37 | ixyd_ | fix or dynamic ip? |
| 10:12.11 | ixyd_ | sounds like getting a new ip over night and having nat/stun issues in the morning ;) |
| 10:13.22 | schmidts | field_it do you see some active channels in the morning? |
| 10:14.37 | field_it | ixyd_: dynamic. yes, that's my guess. but why the heck.. |
| 10:14.45 | field_it | schmidts: nope |
| 10:14.59 | schmidts | ok then it really could be a nat ip problem |
| 10:15.07 | field_it | schmidts: 1st tries fail. dunno yet what triggers it to work again |
| 10:15.27 | schmidts | field_it dns timeout or maybe just reregister |
| 10:15.46 | ixyd_ | i dont have any experiences using external sip providers via nat....so iam not familiar with it :( |
| 10:16.49 | field_it | ixyd_: thanks anyway |
| 10:17.00 | ixyd_ | are you using stun? |
| 10:17.02 | field_it | schmidts: well. yes. sort of. |
| 10:17.07 | field_it | ixyd_: no stun |
| 10:17.08 | ixyd_ | maybe check res_stun_monitor.conf |
| 10:17.10 | ixyd_ | hm :( |
| 10:17.58 | ixyd_ | check the registration interval for your isp |
| 10:18.07 | field_it | hm. how do i force a re-register of sip registries? (w/o reloading entire asterisk.. |
| 10:18.21 | ixyd_ | maybe your ip changes but the registration is not refreshed until your asterisk does so |
| 10:18.37 | schmidts | sip reload |
| 10:18.49 | field_it | ixyd_: could be, looking into it.. |
| 10:18.56 | ixyd_ | good luck |
| 10:18.57 | field_it | schmidts: arg. thx. :) |
| 10:19.32 | field_it | well.. handle_response_register: Forbidden - wrong password on authentication for REGISTER |
| 10:19.40 | field_it | what the .. is going on here |
| 10:20.49 | field_it | oh my |
| 10:21.06 | field_it | ok, this one was also selfmade.. |
| 10:21.18 | field_it | guys, thx and sorry. |
| 10:21.39 | field_it | changed a dyndns registration these days. had a typo |
| 10:21.42 | field_it | but! :d |
| 10:21.57 | field_it | the problem already existed before that! |
| 10:22.56 | field_it | wait. ahhh. it then only started working after resuming the notebook. it previously handled the dyndns update.. |
| 10:23.13 | field_it | yep. so, I'd say, see you tomorrow for confirmation.. |
| 10:23.17 | field_it | ah, another thing. |
| 10:23.27 | field_it | anyone got cdr syslogging working? |
| 10:23.34 | field_it | (presumably yes..) |
| 10:25.53 | field_it | hm, not entirely solved, it seems. |
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| 10:26.54 | field_it | dialling from outside in I get no more "no such number" but plain silence for a couple of secs and then the three short signals in a row. beep-beep-beep (what's it called?) |
| 10:29.40 | field_it | dyndns ip not yet up to date. |
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| 10:33.15 | ixyd_ | field_it: what do you see in your cli while testing? |
| 10:33.53 | field_it | ixyd_: cdr wise? iirc nothing |
| 10:34.12 | ixyd_ | field_it: regarding the 3 beeps |
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| 10:35.07 | field_it | ixyd_: nothing as well. the call doesn't make it through. but that's the not-yet-updated-dyndns-issue I think |
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| 10:37.58 | ixyd_ | ah i see |
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| 10:42.50 | ollii | ehlo |
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| 11:07.31 | skrusty | morning |
| 11:09.08 | skrusty | does anyone know if there is a manger event fired when SIP phones enter DND via feature codes? I know there is a ZAP DND event... |
| 11:10.20 | ixyd_ | i dont know, but u could send one on your own using UserEvent() |
| 11:11.02 | skrusty | yeah |
| 11:11.20 | skrusty | was just hoping there was a standard for it, but never mind |
| 11:11.24 | kaldemar | you'd have to implement the DND feature code for SIP yourself anyway. |
| 11:12.13 | kaldemar | usually DND is a button is SIP phones that just affects the phone behavior and does not interact with the server in any way when activated. |
| 11:12.25 | kaldemar | s/is SIP/in SIP |
| 11:13.57 | skrusty | yeah, i see there's been talk of trying to use options to determin the DND state of a phone |
| 11:14.09 | skrusty | but that doesn't work very well, as most do not implement this correctly it seems |
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| 12:41.44 | tzanger | hm, I'm having a brain fart moment here |
| 12:42.28 | tzanger | three SIP phones calling the same Asterisk server. A calls B, then puts B on hold and calls C. A now does a three-way call with B and C. A hangs up. Does the B-C call leg also terminate? |
| 12:45.48 | ixyd_ | i think there wont be any B-C call-leg |
| 12:46.06 | ixyd_ | as the 3-way conference is done in the phone of A and there a two calllegs A-B and A-C |
| 12:47.44 | ixyd_ | so when A hangs up all regarding calls should be terminated |
| 12:49.33 | leifmadsen | tzanger: is it a conference on the phone itself? |
| 12:49.48 | leifmadsen | if so, then I would suspect the B-C leg would be disconnected |
| 12:55.02 | tzanger | leifmadsen: that's my thinking as well. If the phone's doing the bridging, then the call would very likely drop. If Asterisk was doing the bridge it's less likely |
| 12:55.12 | leifmadsen | yes |
| 12:55.13 | tzanger | I will be trying it shortly but was wondering if there is a "correct" answer :-) |
| 12:55.24 | leifmadsen | the correct answer is to try it :) |
| 12:55.30 | tzanger | leifmadsen: thanks dad |
| 12:55.39 | leifmadsen | tzanger: np grasshopper |
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| 14:07.20 | jaytee | my * box has been hacked |
| 14:09.19 | jaytee | i'm getting outbound call attempts to countries outside the US. The call attempts show up on the CLI and right now my ITSP, Flowroute is blocking them. I don't see any registrations for devices that don't belong. |
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| 14:10.23 | SeRi | jaytee: ouch.... |
| 14:10.49 | ixyd_ | jaytee: allowguest=yes ? |
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| 14:13.41 | jaytee | ixyd, nope it's set to allowguest=no |
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| 14:13.54 | leifmadsen | sounds like either an account with a weak account was compromised, or you've allowed someone with access to [default] dialing out powers |
| 14:14.27 | leifmadsen | should be able to see how those attempts are going out, and thus fix your permissions to block them from doing those calls |
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| 14:16.30 | [TK]D-Fender | Registration is not required to place calls. |
| 14:16.47 | jaytee | when I turned on sip debug I could see the ip address the invite requests were coming from which is in Cairo, Egypt but it's somehow been initiating calls out as if it was one of my internal phones |
| 14:17.08 | leifmadsen | right |
| 14:17.17 | [TK]D-Fender | that would be the "hacked" part. Go lock down your peers, change your passwords and st up something like fail2ban |
| 14:17.18 | leifmadsen | as I said, they likely are using an account with a weak password |
| 14:17.39 | leifmadsen | find out which account they are using, and fix the password -- then fix all your passwords |
| 14:17.52 | leifmadsen | guesses the account names are extension numbers |
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| 14:23.59 | schmidts | leifmadsen :P |
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| 14:44.32 | jaytee | leifmadsen, yep my account names match extensions. looks like I'm going to have to rewrite my dialplan :-( |
| 14:44.45 | jaytee | as well as change all my passwords to something stronger. |
| 14:44.53 | leifmadsen | that's security 101... |
| 14:45.40 | [TK]D-Fender | jaytee, Where are the phones that normally register to those peers located relative to your server? |
| 14:46.05 | jaytee | [TK]D-Fender, they are on my internal LAN |
| 14:46.15 | jaytee | all within less than 100 feet |
| 14:46.20 | [TK]D-Fender | jaytee, then you should ahve had permit/deny to restrict them to your local subnet |
| 14:46.38 | [TK]D-Fender | jaytee, that would have instantly rejected any outside attempt whatsoever |
| 14:46.42 | leifmadsen | still can |
| 14:47.13 | leifmadsen | if your ITSP is also on a static IP, you can also restrict the firewall to only accept incoming calls from those IPs |
| 14:47.15 | jaytee | ok, so I can add permit/deny to each peer entry in sip.conf? |
| 14:47.32 | leifmadsen | jaytee: the better way is to use a template, but yes |
| 14:47.45 | [TK]D-Fender | jaytee, Yes |
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| 14:49.11 | jaytee | that'll help for now and I'm going to put Fail2Ban on this server as well. Trying to think of the best way to "map" my phones to extensions. I have some call macros that are generic for internal extensions. |
| 14:49.33 | leifmadsen | jaytee: i explain a very common way of doing that in asterisk: tdg |
| 14:49.56 | jaytee | mapping the MAC to an extension using the astdb? |
| 14:50.00 | leifmadsen | mac addresses are a good way of naming devices, and using strong passwords -- you can even add a unique identifer on the end to make it a bit stronger |
| 14:50.13 | leifmadsen | jaytee: I use an external DB because it's much easier, but sure |
| 14:51.37 | jaytee | in the meantime, using sip debug allowed me to see the actual source address so I've blocked that in iptables so I'm not getting outbound fraud calls. |
| 14:52.19 | jaytee | definitely time to tighten down this box |
| 14:52.47 | schmidts | jaytee i hope for you it wasnt an too expensive lesson to learn ;) |
| 14:53.09 | jaytee | schmidts, about 15 bucks in calls total |
| 14:53.41 | schmidts | jaytee thats even not enough to really learn something from :D but i am glad for you thats not 15.000 bucks |
| 14:53.45 | jaytee | could have been worse if Flowroute hadn't alerted me this morning |
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| 14:56.07 | chuckf | schmidts: he gets this lesson cheap |
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| 15:23.18 | SeRi | jaytee: indeed. Good luck. |
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| 15:26.04 | vetal | Hi, please help, how can I get ANSWEREDTIME in milliseconds, or something like 1.2 sec& |
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| 15:35.08 | vetal | somobody? |
| 15:35.17 | leifmadsen | if it doesn't so that already, then you would need to change the code to enable that |
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| 15:37.26 | vetal | I can explain for what i need, tarfifcation almost is per second, so if actual duration is 1.2 it need to be rounded to 2 seconds, not mathematical round. But now I see it is mathematical, so I get in cdr from my operator a lot of call that are begger for 1 second |
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| 15:40.13 | tzanger | wow ekiga is a steaming pile of manure on the new ubuntu |
| 15:40.29 | tzanger | what's the preferred gnome sip client these days? |
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| 15:53.14 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:57.33 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 15:59.42 | leifmadsen | tzanger: I like jitsi |
| 15:59.48 | leifmadsen | tzanger: ekiga has always been shite |
| 15:59.51 | leifmadsen | imho |
| 16:03.13 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:03.47 | tzanger | jitsi? never heard of it |
| 16:04.04 | *** part/#asterisk joecool (~joecool@no-sources/joecool) |
| 16:07.49 | leifmadsen | tzanger: yes you have, I just mentioned it |
| 16:09.31 | p3nguin | Wow, a 15$ lesson rather than a $1500 lesson. Well done. |
| 16:10.26 | tzanger | stares blankly at leifmadsen, then takes another sip of coffee, still staring. |
| 16:10.38 | tzanger | p3nguin: nice, what did you learn? |
| 16:10.43 | p3nguin | Not me. |
| 16:12.22 | p3nguin | I understand going in what happens if I don't do things correctly. I take care to make sure this stuff doesn't happen to me. |
| 16:12.58 | *** join/#asterisk cerberus_za (~coert@8ta-151-22-106.telkomadsl.co.za) |
| 16:15.56 | p3nguin | But that's my job, so it should be expected. |
| 16:17.50 | *** join/#asterisk _omer (~omer@182.185.189.121) |
| 16:18.54 | _omer | hello, I deleted some of the LINES from queue_log, now it is stopped getting updated ... May I know why asterisk is not updating queue_log now ? |
| 16:19.18 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 16:20.46 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 16:21.22 | [TK]D-Fender | _omer, Check your permissions on it <- |
| 16:22.10 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:22.56 | _omer | when I do "ls" ... queue_log color is Green .... what does it mean ? |
| 16:23.21 | _omer | I just used chmod +x queue_log and chmod +777 queue_log |
| 16:23.42 | _omer | I am also using queuemetrics ... so how to check permission ? |
| 16:23.52 | p3nguin | Log files do not need to be and should not be executable. It is wrong. |
| 16:24.54 | _omer | sorry I am not good in permission thing. Can you please guide me towards the solution ? |
| 16:25.18 | p3nguin | chmod 0640 queue_log |
| 16:25.32 | tzanger | oh jitsi's a whole communications platform |
| 16:25.51 | ChannelZ | you said you deleted some lines, did you do this with an editor? |
| 16:26.50 | _omer | ChannelZ: yes ... "vi" editor ... then I saved the file and queuemetrics wallboard stopped and queue_log is not getting updated |
| 16:27.03 | _omer | p3nguin: let me check |
| 16:27.15 | ChannelZ | do a "logger reload" on the asterisk console. If it re-wrote a new file asterisk might still have a file handle open to an old node on disk |
| 16:27.40 | ChannelZ | or you might need to restart queues, not sure if its logging goes through the normal channels? hmm |
| 16:28.32 | ChannelZ | off to work.. have fun |
| 16:29.07 | _omer | ChannelZ: I have done "chmod 0640 queue_log" .. let me check if queue_log is getting updated now ... then I will check "logger reload" too |
| 16:31.16 | _omer | still no data in queue_log .... I have issued logger reload as well.... |
| 16:32.14 | _omer | the last line in queue_log is still the same since last 30 minutes ... |
| 16:35.40 | p3nguin | jaytee: This command is good for creating reasonable passwords for (most) phones: apg -a1 -m13 -x26 -MSNCL -E^[]{}:\;\"? -s |
| 16:37.47 | *** join/#asterisk ulogic (~root@ool-4a59e7fc.dyn.optonline.net) |
| 16:40.02 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 16:41.07 | _omer | p3nguin: when I do "ls -l" all files are permitted to " |
| 16:41.11 | _omer | asterisk asterisk |
| 16:41.18 | _omer | but queue_log is permitted to |
| 16:41.20 | _omer | root root |
| 16:41.32 | *** join/#asterisk hfb (~hfb@pool-98-119-109-145.lsanca.dsl-w.verizon.net) |
| 16:41.34 | _omer | I hope you can understand what I mean , I am not good in linux |
| 16:41.49 | p3nguin | chown asterisk asterisk queue_log |
| 16:41.53 | _omer | -rw-r----- 1 root root 11723122 Dec 15 12:44 queue_log |
| 16:41.53 | _omer | -rw-rw---- 1 asterisk asterisk 45617354 Sep 1 21:41 queue_log.0 |
| 16:41.53 | _omer | -rw-rw---- 1 asterisk asterisk 1866007 Jan 30 2011 queue_log.1 |
| 16:42.11 | _omer | ok let me check |
| 16:42.31 | _omer | chown: cannot access `asterisk': No such file or directory |
| 16:43.52 | [TK]D-Fender | ROOT |
| 16:43.56 | [TK]D-Fender | And wrong permissions |
| 16:44.21 | p3nguin | He was able to chmod it, so he's probably root already. I imagine he didn't copy my command. |
| 16:44.31 | pabelanger | your command is missing : |
| 16:44.40 | p3nguin | Oops. |
| 16:44.51 | p3nguin | chown asterisk:asterisk queue_log |
| 16:44.59 | _omer | :) ok let me check |
| 16:46.07 | p3nguin | Now I have to be careful... that fulfilled my mistake quota for the week. |
| 16:46.14 | _omer | works now ..... |
| 16:46.47 | _omer | what does it mean ? -rw-r----- ? all files have -rw-rw---- but queue_log have -rw-r----- |
| 16:47.11 | Dovid | is there any way to see in an agi if a channel is dead or not? |
| 16:47.15 | p3nguin | It means I didn't know what the permissions were on the file before, so I told you a safe permission to use. |
| 16:47.15 | _omer | queue_log file size is still the same where as calls are going thru and file is still not getting updated. |
| 16:48.01 | _omer | ok : let me check logger reload |
| 16:48.04 | [TK]D-Fender | _show us the folder again, and PASTEBIN it this time |
| 16:48.06 | [TK]D-Fender | ~pb |
| 16:48.06 | 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. |
| 16:48.08 | [TK]D-Fender | ^^^^ |
| 16:48.20 | p3nguin | If you feel like changing it to match, which is not going to be necessary in most cases, chmod 0660 queue_log |
| 16:48.48 | p3nguin | 0640 is going to be safe and should still work in almost every case. |
| 16:49.13 | _omer | Great !!! size is getting changed after "logger reload" |
| 16:49.32 | _omer | fffhhheewww .... thanks p3nguin |
| 16:49.40 | p3nguin | What started the problem? |
| 16:49.55 | p3nguin | Did it mess up when you edited the file? |
| 16:50.59 | [TK]D-Fender | yes |
| 16:51.01 | _omer | Let me explain, I edited queue_log file using "vi" then saved and quit ":wq" ..... |
| 16:51.03 | [TK]D-Fender | ^ |
| 16:51.11 | p3nguin | Okay. Don't do that anymore. |
| 16:51.23 | [TK]D-Fender | Stop changing the owners of those files |
| 16:51.39 | _omer | then what is the best way to edit queue_log ? |
| 16:51.48 | p3nguin | You don't need to be editing a LOG file. |
| 16:51.48 | _omer | should I make a copy first ... or what? |
| 16:52.02 | p3nguin | That's the point of having a log file. It records things as they really are. |
| 16:52.21 | _omer | you are right. My editing also disturbed the format. |
| 16:52.31 | p3nguin | Right, so don't do it. |
| 16:52.43 | _omer | I think I should save queue_log in database instead of file ... anyhow. everything looks smooth now .. |
| 16:54.01 | _omer | thanks !! guys :) |
| 16:54.11 | _omer | bye |
| 16:54.15 | _omer | Cheerz |
| 16:58.03 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:00.38 | pigpen | anybody know of any sip issues with asterisk 1.8.3.3 that would cause sip to not "function" |
| 17:00.59 | pigpen | ie: I can see the sip peers, registered, but will not attempt any kind of dial if sip. |
| 17:01.52 | pigpen | ie: dial command sent, but it doesn't |
| 17:02.05 | r0m|u | p3nguin: how is everything working out? |
| 17:03.24 | [TK]D-Fender | pigpen, well you are several releases behind already, but I doubt anything that tragic. you should probably start showing us what you're doing. |
| 17:04.40 | p3nguin | r0m|u: My sound system needs attention. I don't know what to do to it yet. |
| 17:04.58 | r0m|u | p3nguin: alsamixer? |
| 17:05.21 | p3nguin | alsa, yes. alsamixer is just the mixer application. |
| 17:05.37 | r0m|u | I know. by default is muted |
| 17:05.50 | p3nguin | :/ |
| 17:06.00 | p3nguin | This isn't a "default" installation nor a new installation. |
| 17:06.10 | r0m|u | didnt it get upgraded? |
| 17:06.24 | p3nguin | It should have. |
| 17:07.27 | r0m|u | does alse see your card? |
| 17:07.34 | *** part/#asterisk apten (~apten@carbon.gonicus.de) |
| 17:07.44 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 17:08.46 | p3nguin | Of course. I would have no sounds if the card wasn't detected and in use. |
| 17:09.48 | r0m|u | oh. ok I thought you had no sound |
| 17:10.02 | p3nguin | Let me just tell you what the problem is so you don't have to keep telling me things that don't apply. |
| 17:10.14 | r0m|u | :/ |
| 17:10.29 | r0m|u | ill just shut up. |
| 17:10.32 | r0m|u | ;) |
| 17:11.30 | p3nguin | Two things that I can think of right now: tvtime's volume control no longer changes the Line mixer control, and any sounds coming through KDE apps are not obeying the PCM mixer control. |
| 17:12.15 | *** join/#asterisk neurosys (~neurosys@69.198.141.134) |
| 17:12.52 | p3nguin | I keep master and front at 100%. I keep PCM at 50 or less. KDE sounds are full blast (100% volume). |
| 17:13.12 | r0m|u | I see |
| 17:13.27 | p3nguin | I use PCM to adjust my sound level. |
| 17:13.43 | p3nguin | Need more sound, turn up PCM. Need less, turn down PCM. |
| 17:13.53 | p3nguin | Master and front remain at 100% all the time. |
| 17:16.34 | r0m|u | Mhhhh |
| 17:18.32 | p3nguin | It may not be alsa that is the cause of the problems. |
| 17:18.57 | p3nguin | It probably isn't, actually. |
| 17:19.56 | *** join/#asterisk logicwrath (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 17:20.50 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 17:22.25 | *** join/#asterisk c4t3l (~c4t3l@c-76-30-80-232.hsd1.tx.comcast.net) |
| 17:23.25 | r0m|u | I agree |
| 17:25.00 | p3nguin | But it *is* a problem, and I would prefer to fix it soon. |
| 17:25.10 | r0m|u | indeed |
| 17:26.40 | p3nguin | And konqueror is broken. It crashes out within about a minute of my opening it. And the crash handler makes sound, which is full blast out my speakers. |
| 17:31.01 | r0m|u | ouch |
| 17:31.04 | r0m|u | seg fault? |
| 17:32.51 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:33.14 | p3nguin | Yes. sig 11. |
| 17:34.00 | p3nguin | http://pastebin.com/eTrH2pGM |
| 17:37.49 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 17:39.04 | *** join/#asterisk ideaman (~ihaveapla@173-10-29-218-BusName-utah.ut.hfc.comcastbusiness.net) |
| 17:39.27 | r0m|u | looking at it |
| 17:39.34 | ideaman | What is the best channel for some help with some Polycom IP phones and Asterisk? |
| 17:39.41 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 17:39.49 | IsUp | hey |
| 17:40.35 | p3nguin | ideaman: Probably this one. |
| 17:40.51 | ideaman | Alright, well here it goes |
| 17:42.00 | p3nguin | Let 'er rip, tater chip. |
| 17:42.08 | ideaman | I have a TFTP server setup on an Asterisk box for about 10 Polycom phones, all IP650s with older Revs. I've never had to update my bootrom or anything as they have changed Revs. However now, Rev Y phones apparently need a newer bootrom. |
| 17:42.23 | ideaman | My worry that I'm afraid to try without breaking what is currently on the network is... |
| 17:42.25 | r0m|u | lol |
| 17:42.44 | ideaman | Can I just drop a newer bootrom.ld file in there and it'll be backwards compatible? |
| 17:42.45 | p3nguin | If I were you, I would switch to FTP so I can manage my versions by user/pass. |
| 17:43.31 | p3nguin | Don't upgrade your bootrom unless you need to. |
| 17:43.48 | p3nguin | In other words, don't upgrade it on the older phones just for the sake of upgrading it. |
| 17:43.53 | ideaman | right |
| 17:43.55 | ideaman | I don't want to. |
| 17:43.56 | ideaman | but |
| 17:44.12 | p3nguin | FTP, chroot directories, different user/pass for each version. |
| 17:44.29 | p3nguin | Problem solved. |
| 17:44.41 | p3nguin | Polycom phones love FTP anyway. |
| 17:45.48 | anonymouz666 | http://mywiki.wooledge.org/FtpMustDie |
| 17:46.54 | ideaman | Is it hard to convert eveyrthing to FTP if the old ones are all already setup as TFTP |
| 17:47.00 | ideaman | (still learning) |
| 17:48.11 | [TK]D-Fender | ideaman, so you absolutely require an update of the bootrom for something? |
| 17:49.02 | ideaman | I was just thinking that's what my problem was since this one doesn't seem to like the current one in there, and when I asked Polycom, they said this Rev Y needed a newere bootrom |
| 17:49.24 | [TK]D-Fender | Don't put the bootrom in your general provisioning |
| 17:49.39 | [TK]D-Fender | Upgrade just the BR off another FTP folder as a 1-off |
| 17:50.01 | ideaman | I didn't see where you do tell the phone otherwise which bootrom it chooses to load. |
| 17:50.12 | p3nguin | You don't. |
| 17:50.24 | p3nguin | It pulls the one from the directory that it is looking at. |
| 17:50.29 | ideaman | ah |
| 17:50.53 | ideaman | So how can I tell it to look at a different directory. It just repeadly autoboots over and over. |
| 17:50.57 | p3nguin | That's why I said managed directories by using ftp and chroot based on user is the correct solution. |
| 17:51.12 | p3nguin | You don't get to tell the phone that, either. |
| 17:51.46 | p3nguin | You'll upgrade it from a different tftpd or you'll use managed directories and ftp. |
| 17:52.06 | p3nguin | You can even use both -- tftp on the old phones, and ftp on the new ones. |
| 17:52.13 | p3nguin | It's really the best way. |
| 17:52.18 | [TK]D-Fender | Boot the phone go immediately into the menu. hardcode FTP detials. The end |
| 17:52.25 | p3nguin | *nod* |
| 17:52.31 | ideaman | Alright |
| 17:52.34 | ideaman | You guys rock |
| 17:52.51 | p3nguin | Let the ftp server chroot based on the username you enter into the phone. |
| 17:55.54 | r0m|u | p3nguin: I think Qt has a problem with conqueror. |
| 17:56.00 | r0m|u | did Qt get updated? |
| 17:56.13 | r0m|u | konqueror* |
| 17:57.06 | p3nguin | [2011-12-14 12:18] upgraded qt (4.6.3-1 -> 4.7.4-3) |
| 18:02.56 | r0m|u | Mhhhhhhh |
| 18:05.07 | *** join/#asterisk Micc_ (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 18:06.38 | Micc_ | why does asterisk use the internal ip address in the invite and to header when I do just Dial(SIP/user1) but if I do Dial(SIP/1234@user1) it puts the external ip in there and the device ignores the invite? |
| 18:07.34 | Micc_ | do I need to set some kind of nat setting in the peer? |
| 18:07.54 | Micc_ | I already have nat=yes |
| 18:08.42 | IsUp | Micc_: localnet=, externip=, nat=, canreinvite= |
| 18:08.43 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 18:09.15 | Micc_ | asterisk server is not behind nat, but this adtran(user1) is behind nat. |
| 18:10.02 | Micc_ | localnet and externip are global sip settings, not for peer. |
| 18:10.56 | IsUp | ~nat |
| 18:10.56 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:12.34 | Micc_ | I have tons of devices working fine that are just phones. |
| 18:15.21 | Micc_ | I shouldn't need to use externip and localnet if asterisk is on a public ip. |
| 18:16.10 | Micc_ | this seems more like a bug in asterisk to me. why would it be different depending on if I do Dial(SIP/user1) or Dial(SIP/1234@user1) |
| 18:18.02 | Micc_ | I guess I can try sip transforms on the sonicwall and see if that helps. |
| 18:18.06 | kaldemar | Micc_: does the device get the message? SIP/1234@user1 is interpreted as a URI where user1 is a host. |
| 18:18.44 | Micc_ | the device gets the invite but it uses it's external IP in the invite and to headers so the device ignores it |
| 18:18.54 | Micc_ | but it gets it fine when its just SIP/user1 |
| 18:18.57 | kaldemar | Micc_: you only need nat=yes for a peer that is behind a NAT. |
| 18:19.16 | Micc_ | yes the device is behind nat. |
| 18:19.34 | [TK]D-Fender | Micc_, Never allow SIP transfomrs |
| 18:20.08 | [TK]D-Fender | Micc_, show us the attempts with SIP debug enabled along with a peer dump |
| 18:23.59 | voipeng | what breaks it when you use sip transformation? |
| 18:24.14 | voipeng | its just presented incorrectly then? |
| 18:25.37 | Micc_ | http://pastebin.com/th3ra0by |
| 18:26.26 | Micc_ | my scroll back buffer wasn't set enough to get all of the invites on the asterisk side, but you can see them from the peer side. |
| 18:27.08 | Micc_ | is that enough information to see that there is a differencee with the two different ways of dialing? |
| 18:28.28 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:28.34 | [TK]D-Fender | Micc_, that looks like 3rd aprty debug. I want to see what is actually happing at * CLI not after X amount of mangling may have happened along the way |
| 18:29.34 | Micc_ | no mangling, its just behind a comcast modem. |
| 18:30.05 | Micc_ | oh |
| 18:30.12 | Micc_ | my sip show peers says Nat N |
| 18:31.09 | Micc_ | I guess thats what they all say |
| 18:32.02 | Ziaeon | anyone else watching SOPA? |
| 18:32.54 | jaytee | SOPA? on CSPAN? |
| 18:33.00 | [TK]D-Fender | SOPA? NDAA just passed <------- |
| 18:33.10 | jaytee | are they voting today on SOPA |
| 18:33.20 | voipeng | Micc, here are some commands to try on the adtran, used these with support last time deb sip sta messsage |
| 18:33.20 | voipeng | deb sip trunk |
| 18:33.20 | voipeng | deb sip user |
| 18:33.20 | voipeng | deb voice verbose |
| 18:33.28 | jaytee | NDAA pretty much shreds the Constitution |
| 18:33.32 | *** join/#asterisk EugeneKay (eugene@itvends.com) |
| 18:34.02 | voipeng | stack messages is what your looking for i believe, i dont have one here to connect to |
| 18:35.38 | EugeneKay | Not strictly Asterisk, but anybody use voip.ms in concert with CSipSimple(Android) ? I'm having oodles of trouble trying to get calls to work. |
| 18:35.45 | [TK]D-Fender | jaytee, That is the most vile POS I could have ever imagined passing... |
| 18:36.40 | [TK]D-Fender | jaytee, This is the kind of thing that should tip OWS into mass public revolt. You needed a target? You've got one. The police state & military industrial complex |
| 18:37.52 | [TK]D-Fender | jaytee, You have a greater chance of dying in a bathroom accident that due to terrorism in the USA. |
| 18:37.55 | [TK]D-Fender | WAR ON TOILETS! |
| 18:39.06 | [TK]D-Fender | "Those who sacrifice liberty for the sake of safety deserve neither" |
| 18:39.31 | Micc_ | here you go http://pastebin.com/LYS7MvTv |
| 18:40.47 | Micc_ | you see the difference now? |
| 18:40.54 | [TK]D-Fender | Actually no... |
| 18:41.21 | Micc_ | see the first invite after the dial(SIP/waldimports1) |
| 18:41.40 | Micc_ | that has waldimports1@10.1.10.10 in the invite and to header |
| 18:41.47 | [TK]D-Fender | Ah, I see it in the invite header, but not the packet destination or the origin |
| 18:41.53 | [TK]D-Fender | That is odd.. |
| 18:42.03 | Micc_ | yeah those are fine, its just the headers |
| 18:42.18 | [TK]D-Fender | Now try doingt this what we advertise as the "right way" : SIP/peer/number , never sip/humber@peer |
| 18:42.38 | Micc_ | oh, I've just always done it that way for some reason. |
| 18:44.53 | Micc_ | same thing |
| 18:46.54 | [TK]D-Fender | Dopes that device register to *? |
| 18:47.02 | [TK]D-Fender | does* |
| 18:47.27 | Micc_ | yes |
| 18:47.31 | Micc_ | it is registered |
| 18:47.49 | Micc_ | Reg. Contact : sip:waldimports1@10.1.10.10:5060;transport=UDP |
| 18:51.24 | Micc_ | any ideas? |
| 18:51.49 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-uovmthxlwuuwcvtg) |
| 18:51.54 | voipeng | sip inspection on internet router? |
| 18:52.30 | Micc_ | if its a bug, I need some kind of temporary work around. thats why I was thinking transforms might work, but prob not. |
| 18:53.56 | r0m|u | EugeneKay: I use CSipSimple I have not had any issues |
| 18:54.13 | [TK]D-Fender | Micc_, have it register again and validate the contact |
| 18:54.17 | r0m|u | EugeneKay: what type of issues are you running in too? |
| 18:54.29 | EugeneKay | Calls connect, but all I get is silence. |
| 18:54.56 | r0m|u | wireless or over cell? |
| 18:55.03 | EugeneKay | WiFi |
| 18:55.09 | r0m|u | NAT? |
| 18:55.26 | EugeneKay | Yup, though I've tried forwarding UDP:5060/5061 directly to the phone as well |
| 18:55.41 | r0m|u | Did you do the same for RTP? |
| 18:56.03 | EugeneKay | No? |
| 18:56.43 | EugeneKay | What ports should I be fiddling with? This is all still a learning experience for me |
| 19:00.11 | Micc_ | TKD-Fender, you want me to validate by looking at the register packets? |
| 19:00.21 | voipeng | prune peer |
| 19:00.27 | voipeng | register again, reset the device |
| 19:00.28 | voipeng | something |
| 19:01.48 | Micc_ | I see it registering every couple minutes. |
| 19:01.56 | Micc_ | the contact looks the same as in sip show peer |
| 19:02.19 | *** join/#asterisk kikohnl (~kotis@72.253.138.39) |
| 19:02.30 | [TK]D-Fender | Micc_, Yup |
| 19:02.39 | [TK]D-Fender | :/ |
| 19:03.12 | Micc_ | shouldn't asterisk be using the same ip address no matter which way I do the dial? |
| 19:03.32 | [TK]D-Fender | One ould think... This looks tracker-worthy |
| 19:03.43 | Micc_ | seems like an asterisk bug to me, but maybe its supposed to be that way for some reason. |
| 19:04.53 | r0m|u | forward 10000 to 20000 (You could also narrow it |
| 19:05.00 | r0m|u | EugeneKay: ^^ |
| 19:05.22 | r0m|u | you could also do 10000 to 10010 |
| 19:05.35 | EugeneKay | I can do whatever, so long as it works. :-p |
| 19:05.38 | r0m|u | and set csip to use only set rtp |
| 19:05.42 | Micc_ | brb going to try putting behind sonicwall with sip transforms |
| 19:08.27 | EugeneKay | r0m|u - a-ha! I think I found the issue |
| 19:08.40 | EugeneKay | I was previously forwarding 10000 to my brother's desktop for some Steam game of theirs |
| 19:08.45 | r0m|u | EugeneKay: whats that? |
| 19:08.51 | r0m|u | ah! |
| 19:08.54 | r0m|u | ;) |
| 19:09.08 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:09.26 | EugeneKay | And I'm willing to bet csip is just picking hte first port |
| 19:10.17 | r0m|u | EugeneKay: look at the advance settings and you will see wht RTP ports it wnats to use. I set mine to a static port |
| 19:11.11 | EugeneKay | I'm not seeing that in csip, where should I be looking? |
| 19:11.27 | r0m|u | one sec |
| 19:12.21 | EugeneKay | I'd ideally like it to "just work" when I hop on WiFi at a cafe, too, so I'm hesitant to make it static |
| 19:12.38 | EugeneKay | eg, no incoming rewrites needed. |
| 19:13.41 | *** join/#asterisk Micc_ (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 19:13.42 | *** join/#asterisk outtolunc (~outtolunc@c-76-21-78-122.hsd1.ca.comcast.net) |
| 19:13.45 | Micc_ | its working with sip transforms |
| 19:14.00 | r0m|u | EugeneKay: Than you are good leave it as it is. |
| 19:15.28 | r0m|u | EugeneKay: I have to go. work calls. Good Luck. You should be set now!. I might hit you up at #cyanogenmod for some help ;) Take care. |
| 19:16.15 | EugeneKay | Hah :-p |
| 19:16.23 | EugeneKay | Thanks, got me pointed in the right direction |
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| 19:36.03 | Micc_ | now I get an auth reject when I try to make an outbound call with sip transformations |
| 19:43.03 | Micc_ | its not using the username in the invite when making an outbound call with sip transforms. |
| 19:43.10 | Micc_ | and insecure=invite doesn't help |
| 19:43.44 | Micc_ | sip debug doesn't even show the invite packet coming from that ip. |
| 19:44.35 | Micc_ | oh it shows it with sip debug ip |
| 19:45.19 | Micc_ | insecure port fixes it |
| 20:04.08 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
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| 20:45.07 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 20:45.28 | krotos | good evening :) |
| 20:47.23 | idespinner | has anyone ever seen long delays in the AMI between when the originate command is sent and executed and when the extension is actually dialed(up to 10 minutes)? |
| 20:49.00 | krotos | idespinner: no.. |
| 20:49.32 | idespinner | well, figured it was worth a shot atleast :) |
| 20:50.47 | WIMPy | neither |
| 20:51.36 | krotos | idespinner: you use a php-script for AMI-Asterisk? |
| 20:52.02 | idespinner | yes actually. how did you guess? |
| 20:52.11 | idespinner | but its standard TCP/IP sockets |
| 20:52.51 | krotos | okok, yes, directly using tcp sockets. I remember some time ago when i'm was writing my own library for "comunicating" with ast |
| 20:53.08 | krotos | that i had a similar problem, because i not wait the --END-- |
| 20:53.17 | krotos | paste your php code on pastebin |
| 20:53.29 | idespinner | err.. well its pretty big |
| 20:53.58 | krotos | only the crucial parts that comunicate with ast |
| 20:54.01 | idespinner | I actually encapsulated it in a library aswell but I am waiting for the 'so long, thanks for all the fish' clause |
| 20:54.04 | krotos | using ami |
| 20:54.08 | *** join/#asterisk timahvo1 (~rogue@197.176.36.146) |
| 20:54.10 | idespinner | sure |
| 20:54.56 | krotos | idespinner: |
| 20:55.03 | krotos | i paste a simple code for you |
| 20:55.23 | idespinner | i'll just past the whole class.. |
| 20:56.25 | idespinner | krotos, http://pastebin.com/gcjheDDH |
| 20:56.47 | idespinner | the main function is AMI_Originate() |
| 20:56.55 | idespinner | it passes my originate class object... |
| 20:57.35 | krotos | idespinner: http://pastebin.com/ZaYpZkvQ |
| 20:58.00 | idespinner | what is "--END COMMAND--" ? |
| 20:58.08 | *** join/#asterisk asteriskn00b (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 20:58.21 | krotos | response |
| 20:58.23 | krotos | froma st |
| 20:58.26 | krotos | from * |
| 20:58.50 | krotos | it's working for me on 1.8 ast |
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| 21:54.14 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.0 (2011/12/15), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:55.04 | leifmadsen | Both Asterisk 1.8.8.0 and Asterisk 10.0.0 have just been released! Release announcements at http://www.asterisk.org/node/51696 and http://www.asterisk.org/node/51697 |
| 21:55.04 | p3nguin | Happy Birthday, Asterisk 10! |
| 21:57.53 | ponyofdeath | hi, guys is tehre an good wiki or guid on how to secure asterisk? currently i dont have port 5060 open to the outside but would like to do so? |
| 22:03.27 | krotos | happy birthday * 10 |
| 22:04.18 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:06.03 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:07.21 | leifmadsen | krotos: don't you think 10 happy birthdays in a row isn't a bit excessive? |
| 22:07.31 | p3nguin | haha |
| 22:08.11 | The_Boy_Wonder | * 10 for the WIN! |
| 22:08.40 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 22:09.25 | [TK]D-Fender | Apparently he does. |
| 22:09.50 | leifmadsen | I just think it's funny when people type out full words then insist on using * instead of typing asterisk :) |
| 22:09.56 | Micc_ | where is the bug tracker? |
| 22:10.01 | leifmadsen | where isn't it? |
| 22:10.06 | leifmadsen | https://issues.asterisk.org/jira/ |
| 22:10.22 | Micc_ | it used to be bugs.asterisk.org |
| 22:10.30 | leifmadsen | like 3 years ago |
| 22:10.40 | Micc_ | I know, but I still can't remember issues |
| 22:10.51 | leifmadsen | layer 8 problem |
| 22:11.22 | Micc_ | TKD-Fender, any idea what I would search for to find if my bug is already in there? |
| 22:11.47 | [TK]D-Fender | Micc_: Not really... its an odd one |
| 22:12.04 | [TK]D-Fender | Micc_: I'd just as soon post as new and see if a marshall reclassifies it |
| 22:12.05 | Micc_ | yeah, its going to take me a while to search through everything. |
| 22:12.29 | Micc_ | maybe leifmadsen can tell me if he's seen it before? |
| 22:13.03 | leifmadsen | at least try... then when you do file an issue, make sure you provide enough information for someone to reproduce the issue consistently. Provide console output, relevant configuration information, sip traces, and log output |
| 22:13.11 | leifmadsen | Micc_: I don't know what the issue is, so I'll go ahead and say no |
| 22:15.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:16.12 | Micc_ | leifmadsen, when I do a dial(SIP/user1) it sends with the proper internal ip's in the contact and to headers, but when dial(SIP/user1/12345) it send with the external ip in the contact and to headers |
| 22:16.34 | Micc_ | I'm just going to patch it myself if I can find where it is in the code |
| 22:16.54 | *** join/#asterisk kotis_ (~kotis@72.253.138.39) |
| 22:21.30 | *** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4) |
| 22:26.45 | mjordan | Micc_: what version of asterisk? |
| 22:28.50 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 22:29.04 | Micc_ | 1.6.2.20 |
| 22:29.18 | Micc_ | I know its not supported. |
| 22:30.00 | leifmadsen | Micc_: then I wouldn't bother filing an issue unless you can reproduce on 1.8 because that'll be the first thing a bug marshal asks you to do |
| 22:30.52 | paulc | Can I say GotoIf($["${SomeVar}" != "Y" && "${SomeOtherVar}" != "Y"]?somecontext) ? |
| 22:31.01 | paulc | (ie use && as an "and" like that) |
| 22:31.30 | p3nguin | Try using one instead of two. |
| 22:32.34 | p3nguin | I use | for "or" but I don't have any using an "and" like that. |
| 22:33.18 | krotos | leifmadsen: ahahaha, i'm back now , sorry "Happy Birthday Asterisk 10 :-* |
| 22:33.39 | krotos | i'm was busy on reading changelog :) |
| 22:34.16 | paulc | p3nguin: thanks.. I guess "suck it and see" right? I'll give it a whirl and report back.. |
| 22:34.32 | *** join/#asterisk Korolev (~Korolev@nmd.sbx08806.fortlfl.wayport.net) |
| 22:35.21 | p3nguin | I really think you'll end up using & rather than &&, but until someone else says they do it one way or the other, it's only a guess. Try it. |
| 22:39.30 | leifmadsen | p3nguin: 'and' == &, not && |
| 22:39.40 | leifmadsen | errr.... paulc ^^^ |
| 22:40.02 | p3nguin | Now my theory is confirmed. |
| 22:40.12 | leifmadsen | with conditional statements in asterisk, it's just & and |, not && and || |
| 22:40.21 | leifmadsen | and now I'm out |
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| 22:59.59 | Micc_ | I think I'll try 1.8.8.0 right now and see if it solves my problem |
| 23:00.57 | Micc_ | does 1.8.8 support multi-tenant parking? |
| 23:01.12 | Micc_ | that was not in 1.8.5 if I remember |
| 23:11.26 | WIMPy | Ok, so after the new Asterisk releases, when do we get dahdi for the current stable linux? |
| 23:12.07 | p3nguin | You need dahdi for Linux 3.1.something? |
| 23:12.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:14.19 | WIMPy | Wenn, if I have to reboot a system (because I put in some hardware) I use the cahnce to upgrade to a recent kernel, off course. |
| 23:16.43 | SeRi | p3nguin: They replace the addressable Tap on the comcast pedestal in my back yard. |
| 23:16.57 | p3nguin | How's it working out now? |
| 23:17.36 | SeRi | to early to say. Right now I have - uncorrectables so ok for now. |
| 23:17.45 | SeRi | 0* |
| 23:18.10 | SeRi | upstream audio is good as well |
| 23:18.37 | phix | hey gang! |
| 23:18.40 | phix | What's new? |
| 23:19.33 | phix | WIMPy: I usually update hardware too as I usually have awesome uptimes :P when it goes down it is time to upgrade any way |
| 23:19.37 | p3nguin | It was bad before they changed the tap, and it's good after the changed the tap? |
| 23:19.48 | SeRi | p3nguin: yes |
| 23:19.51 | p3nguin | That sounds like success to me. |
| 23:20.00 | SeRi | with comcast success coems witha price. |
| 23:20.09 | SeRi | I rather wait before get my hopes up |
| 23:20.13 | p3nguin | wimpy: I just compiled dahdi 2.5.0.2 on 3.1.5. |
| 23:20.22 | phix | p3nguin: hardcore |
| 23:20.39 | jaytee | livin on the bleeding edge |
| 23:20.59 | *** join/#asterisk Micc_ (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 23:21.05 | phix | Now for your next trick |
| 23:21.17 | Micc_ | and its still a bug in 1.8.8.0 |
| 23:21.17 | p3nguin | I won't use it. I don't use dahdi on my desktop machine. I just did it because it seemed like it was a hard thing to do or something. |
| 23:21.25 | jaytee | please, oh please! let it be warp drive! |
| 23:21.48 | phix | warp, psstt, FTL you mean :P |
| 23:21.55 | phix | You don't want to get too specific |
| 23:21.57 | jaytee | yeah, exactly |
| 23:22.11 | p3nguin | Since it compiled successfully, I don't see a problem with it. |
| 23:22.31 | phix | p3nguin: there are compiler errors and there are runtime errors :) |
| 23:22.41 | p3nguin | I can't test it. |
| 23:23.00 | phix | You know you dont have compiler errors, but a runtime error could still sneak in there |
| 23:23.05 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 23:23.17 | phix | hmmmm why cant you test it? |
| 23:23.17 | p3nguin | Actually, I can test anything that doesn't require special hardware. |
| 23:23.37 | p3nguin | Send me a card that needs dahdi, and I can test that, too. |
| 23:23.39 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 23:27.49 | SeRi | 3.x is not bleeding edge |
| 23:31.23 | p3nguin | Linux 5 is! |
| 23:31.36 | SeRi | ;) |
| 23:32.04 | SeRi | bleeding edge kernels do not make it in to arch or slackware |
| 23:32.32 | SeRi | in slackware you have to compile your own. |
| 23:32.40 | SeRi | I do like teh fact that in arch you dont have too |
| 23:33.27 | [TK]D-Fender | SeRi: http://www.zyra.org.uk/os-air.htm |
| 23:33.34 | [TK]D-Fender | And on that note.. music time, I'm off... |
| 23:34.17 | SeRi | hahaahha! |
| 23:35.20 | phix | p3nguin: :D |
| 23:37.46 | p3nguin | ooooooooooooooh..... |
| 23:37.48 | p3nguin | has a secret |
| 23:37.59 | SeRi | ?????????????????????????????????????????? |
| 23:38.09 | p3nguin | pm |
| 23:51.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-uovmthxlwuuwcvtg) |
| 23:58.16 | *** join/#asterisk `md (yggdrasil@saber.kawaii-shoujo.net) |
| 23:58.48 | `md | hello |