| 00:01.45 | dijib | p3nguin: did you ever solve your issue with MixMonitor sync? |
| 00:01.55 | dijib | and relax ive been drinking |
| 00:02.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 00:03.54 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 00:04.48 | SeRi | waz up guys |
| 00:04.52 | dijib | dude |
| 00:04.56 | dijib | im live |
| 00:05.03 | dijib | at least 36 hours in |
| 00:05.06 | SeRi | dijib: waz up d00d |
| 00:05.11 | SeRi | damn! |
| 00:05.16 | SeRi | no sleep? |
| 00:05.21 | dijib | been drinking. neit. |
| 00:05.25 | SeRi | nice :/ |
| 00:05.29 | SeRi | :) |
| 00:05.29 | dijib | what? |
| 00:05.37 | dijib | us canadians dont work enough |
| 00:05.43 | SeRi | lmao |
| 00:05.44 | dijib | or me, canadian |
| 00:05.56 | SeRi | singular ;) |
| 00:06.02 | ChannelZ | is watching Canada's Worst Driver |
| 00:06.09 | dijib | irregardless. dont say irregardless... those half whit americans |
| 00:06.09 | SeRi | lmao^^ |
| 00:06.21 | SeRi | one sec phone |
| 00:07.41 | dijib | dude i need perl, for wakeup call for asterisk, unless you fine young afro american gentlemen have a like script that i can run though bash php |
| 00:07.55 | dijib | or am i the afro american gentleman |
| 00:08.29 | parasitodelsur | Hu? |
| 00:08.30 | dijib | ChannelZ: are you in ca? |
| 00:08.37 | ChannelZ | no |
| 00:08.49 | dijib | again ive been drinking its most likely my slurry would be understood, likely |
| 00:09.06 | dijib | have you ever driven across black ice? |
| 00:09.29 | ChannelZ | Yes, I live in Colorado |
| 00:09.38 | dijib | dont even know |
| 00:09.42 | dijib | near qb? |
| 00:09.55 | ChannelZ | USA. Colorado. Lots of mountains and snow. |
| 00:10.03 | parasitodelsur | México!? |
| 00:10.26 | dijib | north mid west? |
| 00:10.36 | dijib | not mexico im sure |
| 00:10.56 | parasitodelsur | O ok. I though Colorado was in Mexico. |
| 00:11.20 | ChannelZ | no.. Colorado :) |
| 00:11.47 | ChannelZ | http://maps.google.com/maps?q=golden,+co&hl=en&ll=37.020098,-100.239258&spn=46.288996,46.40625&sll=37.0625,-95.677068&sspn=88.175182,92.8125&vpsrc=6&hnear=Golden,+Jefferson,+Colorado&t=h&z=5 |
| 00:12.26 | parasitodelsur | WoW that's awesome! |
| 00:12.26 | dijib | http://images.4chan.org/k/src/1322438049081.jpg |
| 00:12.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:12.36 | s[X] | hahahaha |
| 00:12.51 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 00:13.33 | [TK]D-Fender | [19:06]dijibirregardless. dont say irregardless... those half whit americans <- "Don't", "wit", "Americans" |
| 00:15.38 | ChannelZ | <sings to the tune Yesterday> Irony... My words actually make fun of me... |
| 00:16.01 | *** join/#asterisk LostyJai (~blah@202.171.190.130) |
| 00:16.16 | LostyJai | hey guys |
| 00:16.22 | LostyJai | in the asterisk full log, does it show what codec is used for each call? |
| 00:17.49 | ChannelZ | No I don't think that's logged by default anywhere |
| 00:18.19 | SeRi | wtf... my system rebooted |
| 00:18.48 | LostyJai | how can i check what codec is used? |
| 00:19.00 | ChannelZ | You can add a Log() something to your dialplan and spit it out |
| 00:19.09 | p3nguin | sip show channels |
| 00:19.11 | LostyJai | we use cisco phones and you can set the "preferred" codec, but that's not necessarily the case? |
| 00:19.15 | LostyJai | good man! |
| 00:19.17 | LostyJai | p3nguin! |
| 00:19.32 | SeRi | waz up p3nguin |
| 00:19.43 | LostyJai | i just see IP and port |
| 00:19.44 | LostyJai | hmmmm |
| 00:19.45 | LostyJai | oh wait |
| 00:19.51 | p3nguin | You can also make sure your phones use the codec you want them to use by disallowing all and allowing only the codec you want it to use. |
| 00:20.01 | LostyJai | the format field? |
| 00:20.22 | LostyJai | in codec.conf? |
| 00:20.30 | p3nguin | sip.conf for sip phones |
| 00:23.26 | SeRi | p3nguin: the light went away for some time here... did you manage to get the file? |
| 00:24.02 | p3nguin | It was reported to be 100%. |
| 00:26.12 | SeRi | p3nguin: cool. I got the the meds. Will be trying it tonight. |
| 00:26.30 | p3nguin | Which dosage did you get? |
| 00:26.35 | SeRi | 3MM |
| 00:26.43 | p3nguin | TR or regular? |
| 00:26.59 | SeRi | regular. they didnt have TR at the store |
| 00:27.03 | SeRi | online only |
| 00:27.08 | p3nguin | That will be a good place to start. |
| 00:27.26 | SeRi | clearence at CVS |
| 00:27.31 | SeRi | 4.87 |
| 00:27.39 | SeRi | 100C |
| 00:27.42 | p3nguin | The purple bottle? |
| 00:27.45 | SeRi | Yes sr |
| 00:27.47 | p3nguin | Cool |
| 00:28.15 | SeRi | I am excited. I want it to work :) |
| 00:28.56 | p3nguin | The only thing I don't like about the kind with B6 in it is that you shouldn't take two pills because they already have 500% of the daily allowance of B6. |
| 00:28.57 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:29.19 | s[X] | fkn drug addicts |
| 00:29.57 | p3nguin | With the ones without B6, you could take two instead if you needed to increase to 6mg. |
| 00:32.08 | SeRi | p3nguin: I see. thats good info. thanks for letting me know |
| 00:32.38 | SeRi | p3nguin: is it possible to set color to the windows tabs on irssi? |
| 00:33.02 | p3nguin | I... don't have any tabs, so I wouldn't know. |
| 00:33.10 | p3nguin | I just have numbered windows. |
| 00:33.23 | s[X] | on what planet is PCI-E the same as PCIx, stupid fkn ebay seller |
| 00:33.38 | SeRi | p3nguin: That |
| 00:33.41 | SeRi | I have the same |
| 00:33.45 | SeRi | I call them tabs :P |
| 00:34.03 | SeRi | 1 2 3 4 5 etc.... |
| 00:34.23 | p3nguin | I suppose you could try to color them, but then the normal colors for activities would be affected. |
| 00:34.56 | SeRi | I see. that makes sense |
| 00:35.52 | p3nguin | I just save my window layout so my channels are always on the same window number. E.g., freenode/#asterisk is always on 30. |
| 00:36.20 | p3nguin | And the keyboard shortcut for that is Alt+x. |
| 00:37.28 | SeRi | cool |
| 00:37.53 | s[X] | You guys on Nix / Win / OSX ? |
| 00:37.58 | p3nguin | Sure. |
| 00:38.09 | p3nguin | "ossix" |
| 00:38.40 | LostyJai | http://pastebin.com/jDFXVMAL |
| 00:38.42 | LostyJai | does this look ok? |
| 00:39.36 | SeRi | p3nguin: lol |
| 00:39.59 | SeRi | s[X]: I am an archaista |
| 00:39.59 | p3nguin | lostyjai: Is that the codecs.conf? |
| 00:40.07 | LostyJai | p3nguin: yes sir |
| 00:40.19 | p3nguin | I use the sample codecs.conf for my stuff. |
| 00:40.20 | s[X] | SeRi..... Dos ? |
| 00:40.30 | p3nguin | If that's what you have, then it's good. |
| 00:40.36 | LostyJai | it's the default |
| 00:41.34 | SeRi | s[X]: lol no. Arch (Arch Linux) aista (user) |
| 00:42.37 | s[X] | I figured u meant archaist and perhaps you used some antiquated operating system lol |
| 00:43.37 | F2Knight | can anyone help me with routing table rules? |
| 00:44.07 | p3nguin | What's the question? |
| 00:44.17 | p3nguin | netstat -nr |
| 00:44.23 | parasitodelsur | I would be very impressed if seri knew such word. |
| 00:44.38 | F2Knight | 2 nic's eth0 and eth1, can not gain proper routing on eth1 |
| 00:44.44 | F2Knight | eth0 works fine |
| 00:44.51 | *** join/#asterisk WiretapNotWorkin (~wiretap@unaffiliated/wiretap) |
| 00:45.08 | p3nguin | Pastebin the output from netstat -nr |
| 00:45.35 | p3nguin | Or be selective and just show me the lines with addresses on them. |
| 00:45.43 | p3nguin | (here) |
| 00:46.17 | F2Knight | http://pastebin.com/R1R8CLfM |
| 00:46.58 | F2Knight | p3nguin, eth1 is a private lan, eth0 is public facing nic. computer does not routing. |
| 00:47.15 | F2Knight | Have been mucking with route and ip route commands trying to fix issue. |
| 00:47.27 | p3nguin | What's line 4 doing? |
| 00:47.44 | F2Knight | it was default from when the system booted |
| 00:47.58 | F2Knight | looks like its just a broadcast |
| 00:48.34 | p3nguin | It appears that it is trying to send all broadcast traffic out eth0. What about the ones that need to go out eth1? |
| 00:48.37 | F2Knight | line 5 i 'think' should read 10.0.0.0 10.126.0.1 255.0.0.0 eth1 |
| 00:49.12 | F2Knight | that is there is a router for the lan at 10.126.0.1 |
| 00:49.29 | p3nguin | 5 is currently only matching traffic to 10.0.0.0, which you probably never use. |
| 00:49.43 | [TK]D-Fender | F2Knight: You may be looking at going to enable IPForwarding in your kernel, and doing NAT for your devices behind * |
| 00:50.06 | p3nguin | Are you trying to use this computer as a gateway? |
| 00:50.11 | [TK]D-Fender | yes |
| 00:50.11 | F2Knight | [TK]D-Fender, trying to NOT do nat.. LAN needs to be a walled garden so to speak |
| 00:50.19 | F2Knight | no not a gateway |
| 00:50.32 | [TK]D-Fender | At least * on both sides... I can only presume he womight want it as a general gatwway as well |
| 00:50.34 | F2Knight | eth1 should be its own independent network from eth0 |
| 00:50.43 | [TK]D-Fender | Ok, or not :) |
| 00:50.48 | p3nguin | So basically it's a dual-homed system in an unconventional way. |
| 00:50.58 | F2Knight | pretty much |
| 00:51.01 | [TK]D-Fender | No, pretty conventional. |
| 00:51.15 | F2Knight | what comes in over eth0 should go out eth0 , what comes on eth1 should go out on eth1 |
| 00:51.15 | p3nguin | Typically a dual-homed system would have a couple public IPs... this has one public and one private. |
| 00:51.32 | [TK]D-Fender | yup |
| 00:51.47 | [TK]D-Fender | Not quite "dual homed" in that sense as I said a hour ot two ago |
| 00:51.47 | dijib | so p3nguin did you ever figure that? or are you going to rely on demux remux for that? |
| 00:51.48 | p3nguin | But it's not a gateway, so that's a bit unusual. |
| 00:52.13 | F2Knight | okay, so its odd.. but how can we make it work? |
| 00:52.14 | dijib | the sync. |
| 00:52.42 | p3nguin | Don't configure NAT. Don't set up ip forward. |
| 00:52.54 | F2Knight | I am sure its a pretty simple route command but ... not sure what to do |
| 00:53.04 | F2Knight | there is no NAT or ip forward |
| 00:53.13 | [TK]D-Fender | othing to set up then |
| 00:53.13 | F2Knight | litterally just added a new nic gave it an IP |
| 00:53.30 | F2Knight | well not [TK]D-Fender because the routing does not work |
| 00:53.31 | p3nguin | You just need a route for each interface, which you have (but you need to fix that one). |
| 00:53.44 | [TK]D-Fender | what "routing"? You are being very vague |
| 00:54.29 | p3nguin | If you send traffic to this system which is destined for the other network which it is connected to, it will route (if you have forwarding enabled). |
| 00:54.33 | [TK]D-Fender | So far there is no "route" between the 2 interfaces. Each is a dead-end to * |
| 00:54.43 | p3nguin | The secret is having a route on the other machines, too. |
| 00:55.08 | p3nguin | Just because you send traffic from one side to a machine on the other side does not mean that receiving one has a route for return traffic. |
| 00:55.27 | p3nguin | That's when you'd implement something like RIP or OSPF. |
| 00:58.15 | p3nguin | Somewhere along the line, I got confused... I thought you didn't want it to be a router, but then you said you did want it to route. |
| 00:58.57 | F2Knight | just restarted the system... |
| 00:59.14 | F2Knight | okay p3nguin I know I need a new route to add to my system. |
| 01:00.01 | F2Knight | I do not want eth1 to 'route' to eth0 ... if it needs the public IP it should go through the gateway device like any other normal lan |
| 01:00.28 | F2Knight | but I do not know what commands to issue to get the gateway route to be correct... gateway not default gateway |
| 01:01.25 | p3nguin | To prevent it from routing, do not enable ip forward at the kernel level. |
| 01:01.41 | F2Knight | which it is not /.. |
| 01:01.47 | [TK]D-Fender | WTF |
| 01:01.58 | F2Knight | but the default route table does not seem to allow proper connectivity |
| 01:02.33 | p3nguin | You should have three routes: one for each directly-connected network, and a default route. |
| 01:02.50 | F2Knight | I mean if I issue a ping -I eth1 10.216.0.1 it should bing the gateway using eth1 |
| 01:03.30 | F2Knight | and it does not . |
| 01:04.17 | p3nguin | route add -net 10.0.0.0 netmask 255.0.0.0 dev eth1 |
| 01:04.43 | [TK]D-Fender | 10.0.0.0 10.126.0.1 255.255.255.255 UGH |
| 01:04.55 | [TK]D-Fender | mask failure |
| 01:04.58 | p3nguin | That should have been configured automatically when you brought up the interface. |
| 01:05.18 | p3nguin | If your interface has 10.x.x.x with a netmask of 255.0.0.0, the route should have been created. |
| 01:06.24 | F2Knight | right but the gateway goes to the default route. |
| 01:06.33 | F2Knight | 0.0.0.0 which is the wrong interfacve |
| 01:06.43 | F2Knight | i need to make the gateway go to 10.126.0.1 |
| 01:06.47 | p3nguin | You want the default gw to use the other interface? |
| 01:07.01 | F2Knight | default gateway for eth0 is fine |
| 01:07.10 | p3nguin | There is only one default. |
| 01:07.15 | F2Knight | but for eth1 it should be set to 10.126.0.1 only |
| 01:07.25 | F2Knight | default yes but not gw |
| 01:07.34 | p3nguin | There is only one default gw. |
| 01:07.45 | F2Knight | default gw = eth0 |
| 01:08.02 | p3nguin | default gw = 216.212.158.1 |
| 01:08.05 | F2Knight | but eth1 needs to go to gateway 10.126.0.1 not the default |
| 01:08.07 | p3nguin | via eth0 |
| 01:08.21 | p3nguin | We're not on the same page, here. |
| 01:08.22 | F2Knight | in other words eth1 needs its OWN gateway |
| 01:08.43 | p3nguin | That's not how routing works. |
| 01:09.45 | F2Knight | thats exactly what it does. ":) |
| 01:10.01 | F2Knight | if a gw is defined it will use that gw. |
| 01:10.06 | p3nguin | No, you're mistaken. There has to be a destination address for the routing table to know where to send the traffic. |
| 01:10.18 | F2Knight | if none are found or 0.0.0.0 is set as the gw it will use the default route |
| 01:10.36 | p3nguin | If you have a route for a network, it will send it to either a next hop or out a device. |
| 01:10.42 | [TK]D-Fender | ^^ |
| 01:10.51 | p3nguin | I've already told you how to send all traffic for 10.0.0.0/8 out eth1. |
| 01:10.53 | [TK]D-Fender | Default is for named subnets |
| 01:10.59 | [TK]D-Fender | or.. NON named oops |
| 01:11.10 | F2Knight | that was already set but it does not work |
| 01:11.14 | p3nguin | It's wrong. |
| 01:11.22 | p3nguin | And I told you how to fix it. |
| 01:11.40 | [TK]D-Fender | bad mask |
| 01:11.45 | *** join/#asterisk _-Jon-_ (~jon@bean.net-xero.com) |
| 01:11.47 | F2Knight | when connecting from a device on the 10 netowrk it does not route back to it because all data is being sent out the default route. I need to define a route for the 10 gateway |
| 01:11.49 | p3nguin | I'd guess the reason it is wrong is because of the wrong mask on the interface. |
| 01:11.59 | p3nguin | Asked and answered. |
| 01:12.09 | p3nguin | Saying it over and over isn't going to change the answer. |
| 01:12.18 | p3nguin | You've been told how to set the route. |
| 01:12.27 | p3nguin | (1904.17) <p3nguin> route add -net 10.0.0.0 netmask 255.0.0.0 dev eth1 |
| 01:12.40 | [TK]D-Fender | Normally that route is automatic when you add the interface |
| 01:12.49 | p3nguin | Yep, but the auto route was wrong. |
| 01:12.57 | p3nguin | Probably due to bad mask on the interface. |
| 01:13.03 | [TK]D-Fender | Which leads me to thinking is ifcfg-eth1 is bad |
| 01:13.04 | F2Knight | up dated route after a reboot http://pastebin.com/pFWXs2GC |
| 01:13.38 | [TK]D-Fender | 169.254.0.0 <- WTF? |
| 01:13.52 | [TK]D-Fender | I don't think you are configuring your NIC's right at all here |
| 01:13.59 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 01:14.00 | p3nguin | APIPA? |
| 01:14.04 | [TK]D-Fender | Auto IP? |
| 01:14.43 | p3nguin | That's probably written in the script to bring up the interface for people who don't know how to otherwise set up networking. |
| 01:14.52 | F2Knight | default configs i suppose |
| 01:15.04 | LostyJai | so i'm calling from my wokr phone to my mobile |
| 01:15.10 | [TK]D-Fender | Configs don't default to knowing what subnet and IP you want for your NIC |
| 01:15.20 | LostyJai | the call drops out on my mobile, but on my work (desktop) phone it remains connected? |
| 01:15.22 | p3nguin | It's fine to have that. It isn't hurting anything as it is. |
| 01:15.27 | [TK]D-Fender | That would require res_psychic.so and I'm not about to hand out my pre-release copy |
| 01:15.31 | F2Knight | thank you p3nguin. |
| 01:15.55 | F2Knight | but when I do something like this ... ping -I eth1 goolge.com it fails |
| 01:15.57 | p3nguin | But now I see on line 7 that you have a valid net destination via eth1. |
| 01:16.36 | F2Knight | yes but an invalid gateway. |
| 01:17.03 | p3nguin | Okay, so we can set a next hop, too. |
| 01:17.09 | F2Knight | line 7 has a gateway of 0.0.0.0 ...the default gateway. ... that goes out over eth0 .. not wha tI want |
| 01:17.20 | F2Knight | okay well this is where I think i got all messed up. |
| 01:17.57 | F2Knight | line 7 : i think should read 10.0.0.0 10.126.0.1 255.0.0.0 eth1 |
| 01:18.23 | [TK]D-Fender | F2Knight: You have a WAN interface. THat is where your default route will go or your internet access on it is fucked. Clear? It needs to be the default route. Your * can talk to whatever other subnets you want, but a GLOBA DEFAULT. |
| 01:18.25 | F2Knight | which would mean this machine would use 10.126.0.1 as the gateway for the 10 network |
| 01:19.24 | F2Knight | [TK]D-Fender, I am not looking to set the default route but assign a gateway route for a specfic network. |
| 01:19.51 | [TK]D-Fender | You should not have to set routes. merely defining your IP properly for your 10 network should do its thing |
| 01:20.03 | [TK]D-Fender | and come up automatically. |
| 01:20.34 | *** part/#asterisk _-Jon-_ (~jon@bean.net-xero.com) |
| 01:21.18 | F2Knight | the problem [TK]D-Fender is that this system (physical machine) can not access the 10.126.0.1 gateway |
| 01:21.24 | F2Knight | because there is no route configured for it |
| 01:21.35 | [TK]D-Fender | ... |
| 01:21.39 | F2Knight | hense why there is needed a route |
| 01:21.40 | [TK]D-Fender | fix your interface definition <- |
| 01:21.50 | WiretapNotWorkin | F2Knight: you don't route to a gateway |
| 01:21.51 | p3nguin | If you ping 10.126.0.1, you get nothing? |
| 01:21.54 | WiretapNotWorkin | you route to a router |
| 01:21.55 | [TK]D-Fender | And what "gateway"? |
| 01:21.57 | WiretapNotWorkin | err |
| 01:22.00 | WiretapNotWorkin | rather, a router routes |
| 01:22.08 | [TK]D-Fender | * is already on the 10 subnet, it doesn't need a "gateway" |
| 01:22.09 | WiretapNotWorkin | you have a default route that goes to the default gateway for your subnet |
| 01:22.23 | WiretapNotWorkin | and you don't need a gateway at all if the devices are on the same subnet |
| 01:23.17 | p3nguin | 10.0.0.0 0.0.0.0 255.0.0.0 U 0 0 0 eth1 <--- this is the route to 10.126.0.1 |
| 01:24.01 | WiretapNotWorkin | yep |
| 01:24.17 | F2Knight | the 10.126.0.1 is a 'router' |
| 01:24.40 | [TK]D-Fender | F2Knight: NO. your box is NOT going out that router. |
| 01:24.45 | p3nguin | And if you send traffic to it which is destined for one of its routes which is not the same network you are on, it will route. |
| 01:24.57 | [TK]D-Fender | F2Knight: Your WAN connected NIC is default, and that's it. ONE default route, and that NIC has to be it |
| 01:25.05 | p3nguin | If you send it to the same network, it will not route. |
| 01:25.39 | F2Knight | if I issue a ping -I eth1 10.126.0.1 I get a response.. I am using eth1 to ping a device. |
| 01:25.58 | [TK]D-Fender | F2Knight: You shouldn't have to specify an interface for ping at all |
| 01:26.12 | [TK]D-Fender | F2Knight: Your box already knows the 10.0.0.0 subnet is on that interface |
| 01:26.23 | F2Knight | if i issue a ping -I eth1 google.com it will not send it to the 10.126.0.1 device to get out |
| 01:26.32 | [TK]D-Fender | OF COURSE NOT |
| 01:26.40 | [TK]D-Fender | You have ONE default route! |
| 01:26.44 | [TK]D-Fender | Not one per interface! |
| 01:26.47 | [TK]D-Fender | ONE |
| 01:26.59 | p3nguin | Set a route to google.com via eth1, and it'll work. |
| 01:27.48 | [TK]D-Fender | F2Knight: Your dream of using that WAN NIC exclusively for phones and using your other router for general traffic is not happening. That is not how this networking works. |
| 01:28.20 | F2Knight | well thats not what I am trying to do [TK]D-Fender |
| 01:28.21 | [TK]D-Fender | F2Knight: You don't get 2 "general" (default) routes out. It doesn't work like that. |
| 01:28.28 | p3nguin | If you configure a route for all applicable networks, it would happen. |
| 01:28.49 | [TK]D-Fender | F2Knight: 10.126.0.1 is not a way for your system to get to google.com |
| 01:29.01 | p3nguin | not unless you tell it to. |
| 01:29.06 | F2Knight | I am trying to keep traffic on the 10 network as if it was a computer with a single nic. |
| 01:29.19 | F2Knight | but that IS the way for that NETWORK to get to the out side world. |
| 01:29.22 | [TK]D-Fender | F2Knight: Your box has a direct internetconnection and is the default route. That is where it will go. |
| 01:29.36 | [TK]D-Fender | [20:29]F2KnightI am trying to keep traffic on the 10 network as if it was a computer with a single nic. <- what part of "not happening" is unclear"? |
| 01:29.50 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:29.56 | F2Knight | the 10 network HAS and USES the 10.126.0.1 gateway/router (little netgear) to get internet access to the 10 network |
| 01:30.04 | p3nguin | Great. |
| 01:30.22 | p3nguin | You want to ping google.com via eth1? Configure a route for it. |
| 01:30.29 | p3nguin | Until then, forget it. |
| 01:30.31 | F2Knight | the box happens to have its own WAN connection for something else all together. |
| 01:30.51 | p3nguin | You chose to have the default gateway on that WAN side. |
| 01:30.54 | F2Knight | so from this box it has a default route over eth0 yes i know it works fine. |
| 01:30.54 | [TK]D-Fender | F2Knight: Since you put a WAN IP on your server directly that is not the case. End of story. That WAN NIC = default and that is the end of it. No more discussion. That is your way to the internet. No secondary choice. No secondary "default route" |
| 01:31.34 | [TK]D-Fender | F2Knight: If you try changing the default route you will kill your WAN NIC effectively |
| 01:31.40 | [TK]D-Fender | ^^^ |
| 01:31.45 | [TK]D-Fender | That is the consequence |
| 01:31.51 | florz | you can perfectly well have multiple default routes active at the same time |
| 01:32.13 | p3nguin | With the use of ip, magical things can happen. |
| 01:32.18 | [TK]D-Fender | Packets might arrive in, but your responses will all go out your LAN based router on another IP and you're screwed |
| 01:32.47 | F2Knight | BUT if from this box I want to check that the LAN side is working correctly , I can not get out side the LAN because the route for eth1 has a gateway of 0.0.0.0 that is on eth0 . If I define the 10 network *on this box only* to use 10.126.0.1 as its gateway, it should route the data to the existing router and work as if it had only one nic and was ascessing the network like anyone else |
| 01:32.53 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 01:32.56 | [TK]D-Fender | Someone talks to you on A. You answer back on B and they say "GTFO" |
| 01:33.35 | *** part/#asterisk SeRi (~ffuentes@c-76-31-169-54.hsd1.tx.comcast.net) |
| 01:33.51 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-xzeimcgbpgxuaxvk) |
| 01:33.51 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 01:33.56 | [TK]D-Fender | ETH0 = outside the LAN |
| 01:34.14 | F2Knight | > sudo route del -net 155.246.75.128 netmask 255.255.255.128 dev eth0 |
| 01:34.15 | F2Knight | > route -n |
| 01:34.15 | F2Knight | Kernel IP routing table |
| 01:34.15 | F2Knight | Destination Gateway Genmask Flags Metric Ref Use Iface |
| 01:34.15 | F2Knight | 192.168.200.0 0.0.0.0 255.255.248.0 U 0 0 0 eth1 |
| 01:34.15 | F2Knight | 155.246.0.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 |
| 01:34.16 | p3nguin | This will be the last time I say it: if you configure a route for the destination with a gateway address of 10.126.0.1, it will work like you expect. Until you create a route for that destination, the traffic is headed out eth0. |
| 01:34.17 | F2Knight | 0.0.0.0 155.246.75.129 0.0.0.0 UG 0 0 0 eth0 |
| 01:34.19 | F2Knight | 0.0.0.0 192.168.200.1 0.0.0.0 UG 100 0 0 eth1 |
| 01:34.20 | p3nguin | f |
| 01:34.21 | F2Knight | 0.0.0.0 155.246.75.129 0.0.0.0 UG 100 0 0 eth0 |
| 01:34.21 | p3nguin | d |
| 01:34.22 | p3nguin | g |
| 01:34.23 | F2Knight | sorry |
| 01:34.56 | [TK]D-Fender | What are those insane WAN routes... |
| 01:35.02 | F2Knight | didnt mean to paste that. |
| 01:35.25 | [TK]D-Fender | Hrm... |
| 01:35.35 | p3nguin | Having two gateways like that won't send your traffic where you intend to send it. |
| 01:35.43 | [TK]D-Fender | Where'd this 200 come from? |
| 01:36.08 | F2Knight | ignore that paste its it not relevent |
| 01:36.12 | p3nguin | There's nothing to distinguish traffic for destination 0.0.0.0 on eth0 from traffic destined for 0.0.0.0 on eth1. |
| 01:37.09 | F2Knight | p3nguin, you said with IP magical things happen .. .are you talking about the 'ip' command from route2? |
| 01:37.22 | p3nguin | But with one having a metric of 0 and one of 100, the 0 is where traffic will go every time. |
| 01:37.27 | p3nguin | Yes, iproute2. |
| 01:37.36 | F2Knight | thats what I thought. |
| 01:37.49 | p3nguin | You can set up multiple gateways with various weights and whatnot. |
| 01:37.57 | F2Knight | yes with ip route i should be able to add a specfic gateway for a network |
| 01:38.21 | p3nguin | You can specify a gateway for a network without iproute2. |
| 01:38.26 | [TK]D-Fender | heads out for a while... |
| 01:38.34 | p3nguin | I've even told you how at least twice. |
| 01:38.53 | F2Knight | it might have gotten lost with [TK]D-Fender telling me its not possible. |
| 01:39.24 | dijib | 20:23 < p3nguin> 10.0.0.0 0.0.0.0 255.0.0.0 U 0 0 0 eth1 <--- this is the route to 10.126.0.1 |
| 01:39.27 | dijib | 20:24 < WiretapNotWorkin> yep |
| 01:39.32 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 01:39.51 | p3nguin | But now he wants to add destinations and gateways. |
| 01:40.01 | florz | F2Knight: well, nearly everything is possible, but without a firm grasp on the concepts it may be difficult to put together anything that works reliably |
| 01:40.03 | F2Knight | correct |
| 01:40.11 | [TK]D-Fender | Only destination we saw was a ping to Google. |
| 01:40.14 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 01:40.34 | p3nguin | And there is no route TO GOOGLE via eth1, only via eth0. |
| 01:40.35 | [TK]D-Fender | Generla internet traffic is not going out that way |
| 01:40.52 | F2Knight | ping to anything outside the 10 network. when using eth1 should route to the 10 networks gateway. 10.126.0.1 |
| 01:40.57 | florz | oh, and BTW "addresses" is orthogonal to "connections" in principle |
| 01:41.05 | p3nguin | Should? No. |
| 01:41.17 | p3nguin | Just because you want it to work that way doesn't mean it *should*. |
| 01:41.31 | [TK]D-Fender | Default route = ETH0 |
| 01:41.57 | p3nguin | Maybe you can explain to him that he still doesn't have a route TO GOOGLE via eth1. I've said it enough and it still hasn't gotten in. |
| 01:41.58 | F2Knight | specficly if i run a program that listens ONLY on one interface. (eth1) and that said program makes a request for connectivity over that networks interface for outside access it should go where? |
| 01:42.13 | [TK]D-Fender | has to be otherwise traffic coming in on ETH0 Will get responded via ETH1 which FUBAR's you |
| 01:42.35 | F2Knight | okay NOT looking to change the default route. |
| 01:42.37 | florz | F2Knight: you cannot "listen on one interface" with IP |
| 01:42.52 | F2Knight | ... |
| 01:42.54 | F2Knight | okay here |
| 01:43.07 | florz | [TK]D-Fender: not necessarily |
| 01:43.16 | F2Knight | tell me how do i check from this system that the 10 network is able to make out side connections using the 10 networks gateway |
| 01:43.17 | [TK]D-Fender | F2Knight: how packets are created is another matter. |
| 01:43.19 | p3nguin | As much as I'd love for you to get the result you want, you won't help yourself, so I'm done. |
| 01:43.52 | p3nguin | |
| 01:44.49 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 01:45.41 | SeRi | :/ |
| 01:46.21 | SeRi | lol |
| 01:52.08 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:52.16 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 01:55.43 | SeRi | p3nguin: you avail? |
| 01:56.33 | p3nguin | Kind of. |
| 01:57.25 | SeRi | I had a questions about irssi but I got it..... I was able to some what modify the numbered screens with a script |
| 01:58.01 | SeRi | My auto reg is not working well.. :/ |
| 01:58.21 | p3nguin | auto... reg? |
| 01:59.02 | SeRi | I mean auto ident |
| 01:59.13 | p3nguin | You're doing it wrong. |
| 01:59.37 | SeRi | i AM SURE LOL |
| 01:59.40 | SeRi | ops caps |
| 01:59.43 | p3nguin | Specify a password for the server. |
| 01:59.44 | SeRi | I am sure :) |
| 02:02.15 | p3nguin | I guess my traffic shaper is killing calls. Randomly, calls hangup and all my SIP registrations drop. |
| 02:05.23 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 02:06.18 | SeRi | p3nguin: how is that? |
| 02:07.49 | *** part/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 02:08.15 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
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| 02:09.52 | SeRi | well I think I messed some shit up on irssi |
| 02:09.58 | SeRi | lol |
| 02:20.54 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-qygsfzufmlffgnkc) |
| 02:20.54 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 02:21.04 | SeRi | lol |
| 02:21.27 | p3nguin | Eh... did that print even before he joined? |
| 02:21.44 | *** join/#asterisk Wiretap_Work (~wiretap@unaffiliated/wiretap) |
| 02:21.53 | SeRi | I dont think do |
| 02:21.57 | p3nguin | It did here. |
| 02:21.58 | SeRi | s/do/so/ |
| 02:22.05 | p3nguin | hahaha |
| 02:22.12 | SeRi | lmao |
| 02:22.14 | SeRi | lol |
| 02:22.21 | SeRi | waht a fuck up |
| 02:22.41 | SeRi | s/waht/what/ |
| 02:22.50 | SeRi | lol |
| 02:24.32 | p3nguin | I think the traffic shaper is causing some problems, and I don't understand why. |
| 02:24.48 | p3nguin | SIP calls and registrations randomly drop. |
| 02:25.11 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 02:25.11 | p3nguin | The registrations usually don't drop alone, but when there is a call, it drops and the regs drop, too. |
| 02:39.41 | *** join/#asterisk x86 (x86@i.am.leet.org) |
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| 03:02.52 | SeRi | Does your shaper have any drop counts? |
| 03:02.58 | SeRi | p3nguin: ^^ |
| 03:03.04 | p3nguin | All 0. |
| 03:03.09 | p3nguin | Puzzling to me. |
| 03:03.16 | SeRi | indeed |
| 03:03.25 | SeRi | messages file? |
| 03:04.52 | p3nguin | What I'd like to do right now is use tshark to make sure packets are being marked correctly. |
| 03:06.12 | SeRi | looks like a good idea |
| 03:07.11 | SeRi | I might drop. working on firewall |
| 03:09.01 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 03:13.05 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
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| 03:18.42 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 03:18.54 | SeRi | looks like I am in now |
| 03:18.55 | SeRi | lol |
| 03:19.48 | *** join/#asterisk n8ideas (~joshelson@216.241.45.82) |
| 03:22.17 | n8ideas | I am looking for a "clean" way to temporarily prevent DTMF tones from being heard in some circumstances... for instance, an agent on an inbound queue call.. anyone done this? |
| 03:23.14 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 03:25.57 | p3nguin | tos 0xb8 |
| 03:26.06 | p3nguin | Which should be decimal 184. |
| 03:26.21 | p3nguin | Which should be DSCF 46. |
| 03:26.32 | p3nguin | Which I hope is codepoint name EF. |
| 03:26.45 | p3nguin | So RTP isn't the problem. |
| 03:26.50 | p3nguin | That was the problem before, but I guess I fixed that. |
| 03:29.37 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
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| 03:37.44 | SeRi | p3nguin: I see. |
| 03:56.14 | SeRi | p3nguin: any progress> |
| 03:56.23 | SeRi | s/>/?/ |
| 03:56.37 | p3nguin | not yet |
| 03:56.59 | SeRi | I see. got an idea of what it is? |
| 03:58.45 | p3nguin | I did learn something, though. |
| 03:59.20 | p3nguin | Wimpy was talking about how the traffic shaping should allow all 100% of bandwidth if nothing else is using it... |
| 03:59.37 | p3nguin | I've learned that Vyatta does that through the ceiling setting in the shaper policy. |
| 04:01.24 | p3nguin | For example, I can set a bandwidth of 2Mbit, and a default class with bandwidth 10% and ceiling 100%, and if other classes are using up their allotted bandwidth, the default class gets 10%. But if the other classes are not using bandwidth, the default class can use up all 100%. |
| 04:01.33 | p3nguin | That's exactly what he was aiming for. |
| 04:01.58 | p3nguin | And it seems that a ceiling of 100% is the default. |
| 04:05.18 | SeRi | pfsense doe it that way.... as soon as it detects voip than it trigers the rules... |
| 04:09.16 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
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| 04:22.16 | p3nguin | What do you think about this policy? http://pastebin.com/V1x9cDhA |
| 04:27.35 | SeRi | p3nguin: seems good to me. |
| 04:27.58 | SeRi | I have been playing with my settings as well. |
| 04:29.36 | *** join/#asterisk BuenGenio (~Gene@180.188.180.24) |
| 04:32.46 | SeRi | p3nguin: let me know if you need to test. I know I do... I want to test my settings.... |
| 04:37.04 | p3nguin | I hadn't had any trouble with it until today, and it dropped every call that was made through my system. |
| 04:37.46 | p3nguin | There's a remote phone which goes through my box to voipms and the inverse as well. |
| 04:38.12 | p3nguin | Every call got dropped after a random time period. |
| 04:38.37 | p3nguin | I've made several calls from my phone on the same lan as asterisk, but none very lengthy. |
| 04:39.09 | SeRi | I see. |
| 04:39.09 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 04:39.16 | p3nguin | And you dialed in to my conf for a long time without ever getting dropped. |
| 04:41.55 | SeRi | Thats true... |
| 04:42.05 | SeRi | Mhhh puzling... |
| 04:45.13 | p3nguin | I didn't touch the vyatta router between yesterday and today, but today calls were fuxed up. |
| 04:45.35 | p3nguin | Do you have a conf up? |
| 04:46.02 | p3nguin | One that I don't have to remember a password for it, and one that I don't get recorded? |
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| 05:05.14 | SeRi | p3nguin: yes one sec |
| 05:11.12 | p3nguin | .win 37 |
| 05:11.22 | p3nguin | Crap, missed again. |
| 05:11.28 | SeRi | lol |
| 05:11.47 | SeRi | now I understand what you mean by that |
| 05:11.48 | SeRi | lol |
| 05:12.26 | SeRi | p3nguin: I gave you the wrong link |
| 05:12.28 | SeRi | look akain |
| 05:12.32 | SeRi | sorry |
| 05:12.39 | SeRi | again* |
| 05:16.40 | *** join/#asterisk freetown (~freetown2@ipvpn098241.netvigator.com) |
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| 06:18.13 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 06:18.25 | SeRi | lol |
| 06:23.17 | irroot | p3nguin what is cisco :P |
| 06:23.56 | p3nguin | Taqua |
| 06:24.00 | irroot | morning folks |
| 06:24.26 | SeRi | g/m |
| 06:24.41 | freetown | real men use procurves |
| 06:24.52 | freetown | or was it D-Link? |
| 06:24.53 | p3nguin | WHAT?! |
| 06:25.58 | p3nguin | MikroTik |
| 06:26.37 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.2) |
| 06:26.43 | freetown | is asterisk 1.4.17 on Hardy known to have problems with Dial? |
| 06:27.00 | freetown | i'm feeding it with the r flag but it won't provide a ringtone... |
| 06:30.53 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-cmwqzhprgxarzown) |
| 06:30.53 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 06:31.00 | irroot | uses tin cans and string |
| 06:31.18 | freetown | bows before the mighty irroot |
| 06:32.07 | irroot | <PROTECTED> |
| 06:32.23 | irroot | folks are worried i might go and get arrested at cop17 |
| 06:32.47 | freetown | irroot, so which are better for long distance? smoke signals or whistle language? |
| 06:33.27 | irroot | freetown best protocol for disemination over long distance is the woman |
| 06:33.49 | freetown | LOL |
| 06:34.19 | irroot | its a rather indescriminate and insecure protocol but has high reliability |
| 06:37.23 | SeRi | dijib: you drunk yet? |
| 06:37.25 | p3nguin | Don't forget to enable the shaper! |
| 06:37.36 | p3nguin | *wink* |
| 06:37.53 | SeRi | core set shit shaper on |
| 06:38.13 | SeRi | dijib: ^^ |
| 06:41.04 | freetown | any suggestions on how I should go about finding out why Dial(,,r) won't provide a ringtone? |
| 06:41.15 | SeRi | are you using a shaper? |
| 06:41.17 | p3nguin | ~r |
| 06:41.17 | infobot | somebody said r was The "r" option to Dial will override any sounds you should be hearing and provide a fake ringing sound to the caller. You generally want the caller to hear the sounds they are supposed to hear, not a fake ringing sound. The caller will hear ringing without the "r" option. Using the "r" option is an edge case and should not normally be used or needed. |
| 06:41.17 | freetown | or Dial(,,m) a music on hold |
| 06:41.24 | p3nguin | ~sipnat |
| 06:41.24 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 06:41.32 | freetown | no natting |
| 06:41.36 | p3nguin | *shrug* |
| 06:41.48 | freetown | <PROTECTED> |
| 06:41.59 | freetown | no shaping either |
| 06:42.09 | freetown | asterisk is supposed to do all the legwork |
| 06:46.18 | *** join/#asterisk irroot (~gregory@197.175.69.203) |
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| 07:20.12 | ChannelZ | Didn't we ponder this the other day? |
| 07:22.40 | freetown | ChannelZ, we did. |
| 07:23.16 | freetown | besides looking at the sip debug scrolling off screen...anything else i can try to find out what is wrong? |
| 07:23.36 | freetown | or is it upgrade your asterisk from that junk that came with Hardy? |
| 07:23.52 | ChannelZ | which is what? |
| 07:24.22 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:24.23 | freetown | Ubuntu 8.04 - asterisk 1.4.17... |
| 07:24.53 | ChannelZ | And this isn't even necessarily an Asterisk problem to begin wtih. If your UCM is answering the channel, then that's all there is to it |
| 07:25.14 | *** join/#asterisk mintos (~mvaliyav@115.184.101.195) |
| 07:25.42 | freetown | yes, but asterisk picks up first from the provider...the people on the line just get silence while asterisk is dialing the extension on ucm |
| 07:26.01 | freetown | it used to provide a ringtone.. |
| 07:26.15 | ChannelZ | And what has changed from then to now? |
| 07:26.34 | freetown | that's the hair tearing part of it...NOTHING. |
| 07:26.47 | ChannelZ | *something* obviously did |
| 07:26.51 | freetown | except maybe a possible upgrade to libraries/kernel/whatever due to the reboot... |
| 07:27.07 | freetown | hates blackouts |
| 07:27.28 | freetown | no configuration changes at all |
| 07:27.55 | ChannelZ | You still haven't really shown anything, to try and narrow the problem down - as I said if the UCM is providing bogus call progress, that's one reason it'd stop "working" |
| 07:28.40 | freetown | the sip debug only shows a sip invite being transmitted to ucm and nothing else... |
| 07:28.53 | freetown | no returns from ucm...unless the call is picked up... |
| 07:29.07 | ChannelZ | so there's your answer |
| 07:29.11 | freetown | i take it that call progress would be packets from ucm? |
| 07:29.34 | freetown | zero call progress? so...why does not the r option take over for me then? |
| 07:29.37 | ChannelZ | It should, at minimum, be replying with a 'Trying' message |
| 07:29.43 | freetown | that's why i had r |
| 07:30.44 | ChannelZ | I'm not sure 'r' works if the other side doesn't even acknowledge the INVITE *at all* and I don't have a good means to test |
| 07:30.56 | freetown | oh... |
| 07:31.39 | freetown | is gonna look up call progress doc on ucm then |
| 07:33.43 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:33.58 | *** join/#asterisk BJD10 (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 07:35.10 | BJD10 | Hi guys, |
| 07:35.11 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:35.37 | BJD10 | how would I go about creating a way of looking up different routes to help choose the lowest cost route. |
| 07:36.09 | ChannelZ | Hmm I just tried dialing a SIP URI that totally doesn't exist and it does ring... so either your UCM is answering the channel or something, or perhaps your indications are borked. |
| 07:36.12 | BJD10 | I have about 15 providers and they all provide a 'prefix' but I do not know how I would make asterisk look at the 'prefix' to compare them |
| 07:37.36 | ChannelZ | BJD10: adding extra logic in the dialplan.. or perhaps faster/easier to write a small AGI script to analyize the dialed number and choose accordingly, as doing it in the dialplan can get messy |
| 07:38.22 | ChannelZ | freetown: is your /etc/asterisk/indications.conf there? |
| 07:43.35 | freetown | ChannelZ, yes...configured by destar...can that mess up ringing? |
| 07:44.56 | p3nguin | You should never let the deathstar configure asterisk. |
| 07:45.26 | freetown | newbie at the time and it purported to be a star for asterisk... |
| 07:45.43 | freetown | so...empty the indications.conf file? |
| 07:46.10 | ChannelZ | freetown: in the case where * is providing the actual tones, indications needs to be configured for the correct country and then the various tones for that country configured below |
| 07:46.14 | BJD10 | ChannelZ: would doing textual searching in an AGI call be expensive? |
| 07:46.36 | ChannelZ | the distributed indications.conf should default to 'country=us' and a related [us] section defining the tone patterns |
| 07:47.02 | freetown | okay...in Hong Kong here...i assume i can find the patterns on voip-info? |
| 07:47.07 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:47.13 | ChannelZ | BJD10: depends on what language you use I suppose. AGI is a relatively simple interface by its self |
| 07:47.42 | ChannelZ | freetown: ?? are you saying your indications.conf file is missing? |
| 07:48.05 | freetown | no...but no HK defined... |
| 07:48.18 | BJD10 | Python I would think would be pretty quick and easy.. |
| 07:48.24 | ChannelZ | freetown: what is country set to? |
| 07:48.28 | ChannelZ | (cn?) |
| 07:48.51 | freetown | us |
| 07:48.51 | BJD10 | but I am confuse as to how I would search for the match. Not the look up in the database.. but the textual matching.. |
| 07:49.14 | ChannelZ | BJD10: I dunno, however one does that in Python... |
| 07:49.23 | BJD10 | I am a little confused as to the 'prefix' ... part; |
| 07:49.44 | ChannelZ | BJD10: me too since I'm not sure what you even mean about the prefix and your 15 providers |
| 07:50.25 | ChannelZ | I'm assuming you mean that certain providers are for certain regions, based on the number being dialed.. which is usually the first few digits (a prefix) that specify regions |
| 07:50.46 | BJD10 | a lot of the providers, have a rate sheet that i can download.. they all have US as '1' for the prefix.. but lots of other prefixes for other countries. |
| 07:51.23 | ChannelZ | well you have to figure out the logic by which you select one provider over another |
| 07:51.24 | BJD10 | so I guess what I am confuesed about is how an international call is routed |
| 07:51.45 | *** join/#asterisk s[X] (~mark@ppp118-208-180-11.lns20.bne4.internode.on.net) |
| 07:51.55 | BJD10 | I have never called an internaitional number so I do not know ... even what they look like |
| 07:51.58 | ChannelZ | freetown: ok that's fine, so long as there is a [us] section also in your indications.. and that Asterisk is reading it properly (it should say as much on a reload) |
| 07:53.07 | freetown | ChannelZ, okay...back to ucm doc then |
| 07:53.16 | ChannelZ | BJD10: depends on what country you're in. From the US for instance you dial 011, then a country code like 44 for the UK, then the number... |
| 07:54.27 | ChannelZ | freetown: I mean the proper way to do it is the UCM to be providing real call progress based on whatever it's doing rather than trying to patch around it with 'r' on Asterisk |
| 07:55.18 | ChannelZ | freetown: but it's always possible there is some odd bug in * in your case, your package upgraded and screwed something up, who knows |
| 07:55.25 | freetown | ChannelZ, i understand what you are saying...iirc...you did say that ucm has a knack of not providing call progress? |
| 07:55.42 | ChannelZ | freetown: no, I don't know a single thing about this UCM |
| 07:55.59 | ChannelZ | If what you're saying is true however, that it's not replying to SIP packets *at all*, it certainly isn't providing progress |
| 07:56.18 | ChannelZ | I'm not actually sure how it's working at all to be honest if this is really the case |
| 07:56.21 | freetown | ucm? cisco unified communications manager...the cisco ipphone/phone solution... |
| 07:56.50 | ChannelZ | No, I know what it is, but I mean I've never used one, have any experience with one, or any other knowledge of them |
| 07:57.15 | ChannelZ | Alas from the outside looking in, it doesn't seem to do SIP very well. |
| 07:57.36 | ChannelZ | Not sure if that counts as a "phone solution" :) |
| 07:58.31 | freetown | reading up on docs right now. thanks for the pointers ChannelZ |
| 07:58.52 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 07:59.27 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:59.36 | ChannelZ | yeah well good luck, it's hard to diagnose a secondary system interaction without direct experience with the system in question or even seeing any debug output |
| 07:59.55 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:01.08 | BJD10 | ChannelZ: so if dialing from US to the UK I would dial 011 + 44 + there number? |
| 08:01.21 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:01.23 | schmidts | good morning |
| 08:02.57 | ChannelZ | BJD10: yes, typically.. your various ITSPs might have slightly different forms they want you to dial, possibly. |
| 08:03.55 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:04.02 | ChannelZ | morning schmidts |
| 08:04.13 | BJD10 | ChannelZ: I guess there is no rule as to how long the 'phone number' portion is is there |
| 08:05.36 | ChannelZ | I'm sure there is, I just don't know what it happens to be |
| 08:05.52 | ChannelZ | And different countries surely have different dialplans |
| 08:06.49 | p3nguin | Some of those countries have variable length phone numbers, so it could get messy if you're trying to match the entire number. |
| 08:07.19 | BJD10 | p3nguin: just trying to figure out how to determing the 'prefix' correctly for routing |
| 08:07.29 | p3nguin | What's a prefix? |
| 08:08.03 | ChannelZ | thanks for driving this car in a circle |
| 08:08.14 | p3nguin | :/ |
| 08:08.38 | kaldemar | it's the opposite of suffix. |
| 08:08.49 | p3nguin | Oh, no shit? |
| 08:08.56 | ChannelZ | BJD10: are you indeed dialing out of the US? |
| 08:09.23 | p3nguin | If you're talking about adding a 1 for US numbers or 011 for everything else, I guess it seems pretty clear how you'd know which one you're dialing. |
| 08:12.20 | irroot | irony as climate confrence opens in durban south africa there are 8 killed in floods in durban ... on the agenda is climate change droughts in africa |
| 08:14.26 | ChannelZ | Al Gore's wet dream |
| 08:14.42 | p3nguin | Anyone here using squareup? |
| 08:15.18 | p3nguin | Square/squareup.com... whatever |
| 08:15.28 | ChannelZ | the credit card thingy? |
| 08:15.31 | p3nguin | yes |
| 08:16.02 | p3nguin | I'm wondering if it's my imagination that the fee went up. |
| 08:17.24 | ChannelZ | hmm dunno. I don't use it, just know some who do |
| 08:17.35 | *** join/#asterisk pietro1 (~pietro@88-149-227-51.dynamic.ngi.it) |
| 08:18.38 | ChannelZ | a random article says 2.75% + $.15, dated November 2010 |
| 08:19.18 | BJD10 | ChannelZ: well yes I am in the US :) and while I mainly care about the US routes from the providers If I can define how to sort the prefixes then I can enable INT calling as well |
| 08:19.19 | ChannelZ | seems to be the same as what their website says now, minus the $.15 |
| 08:19.42 | ChannelZ | BJD10: well a US prefix (the area code I think you mean) is the first three numbers |
| 08:19.51 | p3nguin | I guess it went down then instead of up. |
| 08:20.10 | ChannelZ | 3.5% + $.15 if you enter CC numbers manually. |
| 08:20.18 | p3nguin | oh |
| 08:20.27 | ChannelZ | https://squareup.com/pricing |
| 08:20.44 | BJD10 | ChannelZ: thats our area codes.. but the US prefix is 1 on all the lists |
| 08:21.01 | p3nguin | And everything else will be something else. |
| 08:21.14 | ChannelZ | well yes, usually you have to dial 1 to make a 'long distance' call |
| 08:21.32 | ChannelZ | on POTS. My ITSP, Vitelity, always requires the 1 even for local calls |
| 08:21.32 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 08:21.39 | BJD10 | btw. I use square up.. 2.75% on swiped cards no per transaction fees |
| 08:21.41 | ChannelZ | 011 is for international |
| 08:21.52 | BJD10 | 3.something on non swiped cards |
| 08:21.55 | ChannelZ | so maybe that's all you're asking. I really don't know. |
| 08:22.26 | p3nguin | http://www.countrycodes.com/ |
| 08:22.33 | BJD10 | so from what I am getting it is like this ... <countrycode>+<areacode> is = prefix |
| 08:22.42 | BJD10 | except the US is just 1/ |
| 08:23.04 | p3nguin | 1 is the country code. |
| 08:23.11 | ChannelZ | yes |
| 08:23.12 | BJD10 | seems the UK listings are all 44+ a few digits.. like 447939 |
| 08:23.22 | p3nguin | area code is not a prefix, it's part of the phone number. |
| 08:23.26 | BJD10 | or 44118005 |
| 08:23.27 | ChannelZ | right.. |
| 08:23.45 | BJD10 | So I am guessing that when in the uk , you would not dial the 44 part |
| 08:23.51 | p3nguin | The UK area code is 44. |
| 08:24.09 | irroot | US / Canada 1 Russia is 7 i think here in south africa its 27 so +2711 is Johannesburg 11 area code the + translates in most places to 00 [int access] |
| 08:24.17 | BJD10 | and just the 7939.... or the 118005.... both of them having part of the areacode and a number in it |
| 08:24.28 | p3nguin | For an international call, we use a 011 access code, the 44 country code, the area code, and the rest of the number. |
| 08:24.33 | s[X] | Hey irroot u from SA ? |
| 08:24.41 | irroot | yip jozi |
| 08:24.45 | s[X] | lekker |
| 08:24.57 | irroot | poes lekker |
| 08:25.01 | s[X] | :P |
| 08:25.07 | irroot | s[x] where you at |
| 08:25.09 | s[X] | im from Empangeni |
| 08:25.12 | BJD10 | irroot: my list is showing 2711 as Johannesburg |
| 08:25.18 | freetown | ChannelZ, http://pastebin.centos.org/38084 |
| 08:25.25 | s[X] | 2711 is joburg |
| 08:25.36 | freetown | the entirety of the sip conversation between asterisk and ucm |
| 08:25.43 | freetown | ucm actually did send a 200.... |
| 08:25.48 | irroot | Empangeni see the COP17 fools brought some flooding your side |
| 08:26.00 | s[X] | Im in Australia now :P |
| 08:26.04 | BJD10 | but yes all the SA seems to start at 27 and some numbers... so irroot how would you dial a local number? |
| 08:26.08 | p3nguin | 011 or 00 are acceptable access codes. |
| 08:26.08 | irroot | s[x] ah |
| 08:26.13 | ChannelZ | freetown: can't be, there's not even an INVITE in that. |
| 08:26.57 | freetown | hmm...maybe sip set debug ip ain't good enough then... |
| 08:27.05 | irroot | BJD10 if i was to dial a local number my local access code is 0 so i dial 011XXXXXXX for johannesburg and 012XXXXXXX for pretoria |
| 08:27.06 | freetown | i guess i need it more verbose... |
| 08:27.19 | s[X] | South African introduced full 10 digit dialing |
| 08:27.22 | irroot | to call london it be 0044207XXXXXXX |
| 08:27.23 | s[X] | South Africa* |
| 08:27.37 | s[X] | Yeah int outbound dialing code for sa is 00 |
| 08:27.40 | ChannelZ | It should be. You turned on debug and THEN made a test call? |
| 08:27.47 | s[X] | 00 + Country Code + Number |
| 08:27.52 | irroot | s[X] yeah got rid of all the party lines and now on fully automated exchanges |
| 08:27.54 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:27.55 | ChannelZ | There should be a crapload of packets |
| 08:27.59 | BJD10 | okay so the 0 is your areacode... what is the 11 part or the 12 part? |
| 08:28.02 | p3nguin | 00 is the access code, 44 is the country code, 207 is the area code... |
| 08:28.15 | s[X] | i left 10 years ago hehe |
| 08:28.15 | p3nguin | 0 is the access code, not area code. |
| 08:28.24 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 08:28.26 | BJD10 | oh okay |
| 08:28.31 | irroot | BJD10 0 is local access 00 is international access area code is 11 |
| 08:28.31 | p3nguin | 27 is the area code. |
| 08:28.35 | BJD10 | are area codes across the global 3 digits? |
| 08:28.37 | p3nguin | But it's local, so it is left out. |
| 08:28.48 | irroot | p3nguin 27 is country code |
| 08:28.52 | p3nguin | sorry |
| 08:29.09 | freetown | ChannelZ, yeah...i turned on debug and then made a test call... |
| 08:29.31 | ChannelZ | on the right system!? :) |
| 08:29.38 | freetown | i could have the phone on the ucm side of things ringing away while the cell just gave me silences |
| 08:29.47 | freetown | s/silences/silence/ |
| 08:30.07 | singler | BJD10: no, in my country area codes varies from 1 to 3 digits |
| 08:30.39 | BJD10 | Think I got it.. from the US we have to dial 011 to get international access. Other wise its considered local, or a 1 for long distance/local |
| 08:30.44 | ChannelZ | freetown: something is seriously hosed, you should have gotten an incoming INVITE from your ITSP and some additional chatter, as well as the outgoing INVITE to your UCM and subsequent chatter |
| 08:30.56 | ChannelZ | you did "sip set debug on" ? |
| 08:31.04 | p3nguin | I think UK area codes are 2-4 digits. |
| 08:31.15 | freetown | just did...but then i get way too much chatter on screen... |
| 08:31.24 | ChannelZ | well yeah |
| 08:31.32 | ChannelZ | so wtf did you paste then |
| 08:31.39 | freetown | is it possible to tie things down to one particular sip line? |
| 08:31.52 | ChannelZ | you can do it by ip |
| 08:31.53 | freetown | then i can filter out the additional chatter |
| 08:31.57 | freetown | -_- |
| 08:32.06 | freetown | same blooming ip for all lines :D |
| 08:32.09 | ChannelZ | sip set debug ip x.x.x.x |
| 08:32.19 | ChannelZ | Well life sucks |
| 08:32.27 | irroot | <PROTECTED> |
| 08:32.28 | freetown | it does... |
| 08:32.40 | BJD10 | so anything after 011 I can strip for the prefix to route with... the question then is if I use an agi script to query a database of prefixes.. how much lag will searching for it result in... |
| 08:32.45 | irroot | +27 87 9409936 is sip on my netbook |
| 08:32.54 | p3nguin | five words: test en vi ron ment |
| 08:33.00 | singler | freetown: ues tcpdump to capture sip, and then use wireshark to analyze voip calls, you'll get nice graphs |
| 08:33.04 | ChannelZ | BJD10: depends how big your database is... what it is... |
| 08:33.13 | irroot | its a NGN the "areacode" 87 is reserved for VOIP termination |
| 08:33.17 | s[X] | ArchLinux is taking its sweet ass time to install |
| 08:33.29 | ChannelZ | It's probably going to take a second or less even with a huge database |
| 08:35.20 | ChannelZ | freetown: let's start very simple. do "core set verbose 5" and then make a test call, pastebin that. |
| 08:35.49 | ChannelZ | let's see what Asterisk thinks is happening |
| 08:36.08 | irroot | BJD10 i have a database of international codes also some prefix descriptions |
| 08:36.28 | BJD10 | irroot: how do you select the route to use to dial ? |
| 08:37.11 | freetown | ChannelZ, got a dump here: http://pastebin.centos.org/38085 |
| 08:37.30 | freetown | i just realized that i could have a better filter with the other ucm ip |
| 08:37.32 | freetown | trying |
| 08:37.35 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:37.37 | BJD10 | here is an example of the rates download http://voip.ms/rates.php about 10K records, I have several lists like this about 100K records |
| 08:37.45 | irroot | BJD10 i have a AGI script that does this also calculates the cost and alocates the cost |
| 08:38.46 | ChannelZ | freetown: well if 10.1.2.1 is your UCM, it seems to be doing progress OK.. it replies with "Trying", and even "Ringing" |
| 08:39.42 | ChannelZ | in which case.. have you even tried without 'r'? |
| 08:40.27 | BJD10 | irroot: is this a script you wrote? |
| 08:42.01 | irroot | based on A2Billing but complete hatchet job |
| 08:43.20 | *** join/#asterisk BuenGenio (~Gene@218.189.219.210) |
| 08:44.55 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 08:47.56 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 08:48.04 | giany | how can i disable the code for #1 ? |
| 08:48.42 | ChannelZ | giany: features.conf |
| 08:50.52 | giany | that is what I thought too.. |
| 08:50.54 | giany | thx |
| 08:50.56 | ChannelZ | assuming your'e even using features I guess. Otherwise it's possibly something your device is doing |
| 08:51.12 | giany | in the Dial app what do tTW mean? |
| 08:51.44 | ChannelZ | t and T allow transferring by either the called or calling party respectively.. so you could remove those too |
| 08:51.56 | ChannelZ | W is recording |
| 08:52.06 | ChannelZ | ack |
| 08:52.14 | giany | thx |
| 08:52.16 | ChannelZ | W is recording by the calling party |
| 08:52.29 | ChannelZ | s/recording/allow recording/ |
| 08:52.40 | ChannelZ | should go to bed |
| 08:52.45 | *** join/#asterisk irroot (~gregory@197.111.202.166) |
| 08:52.52 | giany | ok, thx its clear |
| 08:56.18 | freetown | ChannelZ, I have tried with tT only...no joy... |
| 08:56.19 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:58.33 | ChannelZ | freetown: I don't know why. You said you're calling from a cell? |
| 08:59.03 | freetown | only thing available to make a call to the sip line(s) |
| 08:59.36 | ChannelZ | so you haven't tried a softphone somewhere else? |
| 09:00.05 | ChannelZ | I can try calling you if you want |
| 09:00.27 | ChannelZ | or wait.. what country are you in |
| 09:00.59 | freetown | ChannelZ, i have a softphone to asterisk...using that to call an extension on ucm works just fine... |
| 09:01.02 | *** join/#asterisk Nasga (~Nasga@110.113.117.78.rev.sfr.net) |
| 09:01.08 | ChannelZ | you hear ringing |
| 09:01.12 | freetown | yes |
| 09:01.52 | freetown | you reckon the nortel crap from the sip provider is at fault now? |
| 09:02.17 | ChannelZ | ok well that would have been nice to know. It might be something screwball with your cell provider |
| 09:02.42 | freetown | it's not just cell...even landlines.... |
| 09:02.43 | ChannelZ | Again I haven't even seen your dialplan and how/why/what asterisk is doing in the middle of all this |
| 09:02.59 | freetown | i'd be happy to post the dialplan for you |
| 09:03.14 | ChannelZ | well I was waiting to see your verbose output which will tell a lot |
| 09:03.37 | freetown | oh that. coming |
| 09:07.54 | freetown | ChannelZ, http://pastebin.centos.org/38086 |
| 09:08.29 | freetown | oh...should include sip stuff for asterisk <-> provider too? |
| 09:10.47 | *** join/#asterisk pietro1 (~pietro@88-149-227-51.dynamic.ngi.it) |
| 09:11.05 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 09:11.11 | ChannelZ | Did you not answer the call on the UCM side and just hang up? |
| 09:11.48 | freetown | No, I did not answer the call on the ucm side...i just hung up after four rings or so |
| 09:11.54 | ChannelZ | ok |
| 09:11.55 | freetown | hung up the cell that is |
| 09:12.35 | freetown | zero natting...asterisk is the go between from ucm and provider... |
| 09:12.38 | ChannelZ | well it's reporting ringing, so either it's not generating the sound, or it's getting muted for some reason up the line (your ITSP or the cell) |
| 09:13.07 | freetown | muted? just the ringing only? |
| 09:13.54 | ChannelZ | why with 'r' it's not generating ringing either also seems like it's getting muted elsewhere.. unless your Asterisk has become screwed up and it is unable to generate the tones either which I can't guess how that can even happen if your indications.conf is default |
| 09:14.23 | freetown | indications has been touched by destar... |
| 09:14.54 | ChannelZ | I don't think the communication between your ITSP and Asterisk is a factor since you're Answer()ing the channel on the Asterisk side. |
| 09:15.04 | ChannelZ | destar? |
| 09:15.25 | freetown | asterisk configuration software with big fancy claims |
| 09:15.42 | ChannelZ | oh.. another GUI? these things are like infections |
| 09:15.54 | freetown | if only i had known... |
| 09:16.12 | ChannelZ | so touched how? |
| 09:16.15 | freetown | http://www.voip-info.org/wiki/index.php?page_id=134&comments_page=1 |
| 09:16.23 | freetown | should i give that hk indications a go? |
| 09:16.29 | irroot | ChannelZ lol so what that make me i supply asterisk with Gui |
| 09:16.41 | freetown | ; Automatically created by DESTAR |
| 09:16.51 | freetown | that's at the top of the indications file |
| 09:17.06 | freetown | i have no idea how it differs from default |
| 09:17.47 | *** part/#asterisk pietro1 (~pietro@88-149-227-51.dynamic.ngi.it) |
| 09:17.49 | irroot | freetown download asterisk source in there are the default configs |
| 09:18.06 | freetown | k |
| 09:18.07 | ChannelZ | http://pastebin.com/tvEYHdv9 |
| 09:18.07 | *** join/#asterisk mandla (~quassel@168.167.180.161) |
| 09:18.12 | ChannelZ | make it that |
| 09:18.17 | mandla | Morning... |
| 09:18.41 | freetown | ChannelZ, what about the hk indications on that voip-info page? |
| 09:19.58 | ChannelZ | whatever, you can try it though their syntax isn't quite right |
| 09:20.58 | freetown | right...i'll blow indications away and stick the default in then |
| 09:21.02 | ChannelZ | makes wierd noises here |
| 09:21.12 | freetown | HK is weird :p |
| 09:21.29 | ChannelZ | it made like half a US ring and then just a long tone forever |
| 09:22.09 | freetown | ooh...there was an indications.conf.orig... |
| 09:22.11 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 09:22.13 | freetown | stuck that back in |
| 09:26.11 | *** join/#asterisk s[X] (~mark@ppp118-208-122-13.lns20.bne4.internode.on.net) |
| 09:27.53 | freetown | no joy. |
| 09:30.16 | ChannelZ | with r? |
| 09:30.23 | freetown | with r |
| 09:31.44 | ChannelZ | has no idea |
| 09:32.17 | freetown | ChannelZ, but thank you for your time and effort here. Really appreciated |
| 09:33.25 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 09:33.36 | hrolf | How do I get the call start time in dial plan? |
| 09:34.38 | *** join/#asterisk datalay (~datalay@unaffiliated/datalay) |
| 09:35.05 | kaldemar | hrolf: CDR(start) |
| 09:36.32 | hrolf | kaldemar: Thanks. |
| 09:38.10 | ChannelZ | freetown: can you make a test extension that calls PlayTones("440+480/400,0/200,440+480/400,0/4000") and then a Wait(10) and then call it? |
| 09:39.55 | freetown | k |
| 09:42.03 | ChannelZ | (oh.. and Answer() first) |
| 09:46.21 | *** join/#asterisk irroot (~gregory@197.109.50.234) |
| 09:46.26 | freetown | ChannelZ, oh...a sip extension? |
| 09:46.43 | freetown | or something magic in extension.conf? |
| 09:48.05 | ChannelZ | oh.. well I guess you'd have to hotwire your 's' to duplicate your normal calls.. but I guess this is a running system doing legitimate things |
| 09:48.18 | freetown | res_indications.c:212 handle_playtones: Unable to start playtones |
| 09:48.23 | *** join/#asterisk irroot (~gregory@197.104.137.168) |
| 09:48.55 | ChannelZ | hmmm interesting |
| 09:49.07 | freetown | that's what asterisk reported after PlayTones got triggered |
| 09:49.23 | ChannelZ | try "module load app_playtones" on the console |
| 09:49.25 | freetown | <PROTECTED> |
| 09:49.26 | ChannelZ | then try again |
| 09:49.45 | freetown | Module 'app_playtones' could not be loaded |
| 09:50.01 | freetown | don't tell me that is necessary for Dial r |
| 09:50.06 | wdoekes2 | res_indications ? |
| 09:50.37 | ChannelZ | res_indications is built-in. app_playtones is just the dialplan app PlayTones() |
| 09:51.05 | ChannelZ | I guess maybe your package doesn't have it built. grrph. |
| 09:51.18 | freetown | oh. so Dial r should work? ??? |
| 09:51.49 | ChannelZ | ideally yes but I'm not sure why it isn't.. even assuming it isn't |
| 09:52.03 | wdoekes2 | Dial r needs the indications, it shouldn't need the playtones app |
| 09:52.04 | ChannelZ | this could be upstream from you |
| 09:53.20 | ChannelZ | although hmmm |
| 09:55.22 | *** join/#asterisk zogg_laptop (~michael@213.8.57.217) |
| 09:55.27 | zogg_laptop | hey |
| 09:55.58 | zogg_laptop | i compiled dahdi on centos but as i try service dahdi start it says no dahdi service, did i miss anything? |
| 09:55.59 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.2) |
| 09:56.32 | ChannelZ | the start scripts aren't installed by default if I remember right |
| 09:56.37 | ChannelZ | init scripts rather |
| 09:57.13 | zogg_laptop | ChannelZ, how do i install it than? |
| 09:57.43 | zogg_laptop | i think i did compiled somewhere dahdi before and it actually did install the script |
| 09:58.15 | ChannelZ | I think they're part of dahdi-tools |
| 09:59.04 | ChannelZ | and even then it's "make config" to install them, or you do it yourself |
| 09:59.19 | zogg_laptop | i got dahdi-linux-complete-2.5.0.2+2.5.0.2 so i asume it has tools, or might be wrong?\ |
| 09:59.50 | zogg_laptop | ChannelZ, i didn't do make config as i use my configs |
| 10:00.02 | freetown | ChannelZ, with the m option, the softphone gets music... |
| 10:00.14 | ChannelZ | freetown: something odd is going on looking at the source based on that error you posted above, but I really have to go to bed. Maybe your * install IS jacked. |
| 10:00.26 | freetown | jacked? |
| 10:00.33 | freetown | broken? |
| 10:00.43 | ChannelZ | zogg_laptop: make config in dahdi-tools, not Asterisk. |
| 10:00.47 | freetown | ChannelZ, thanks for all the help |
| 10:00.56 | freetown | i have to go too. |
| 10:01.26 | ChannelZ | freetown: the 'Can't start playtones' or whatever it was is unrelated to the PlayTones app, but I'm not sure why it's unable to play. |
| 10:02.12 | ChannelZ | Your package probably has all the dialplan apps compiled in not as modules which is why the module load returned an error; if you do "core show application playtones" it probably shows you the app help correctly |
| 10:02.34 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 10:03.36 | zogg_laptop | ChannelZ thanks =) |
| 10:03.39 | ChannelZ | you know what i lied to you, I put quotes around that PlayTones string and there shouldn't be |
| 10:03.47 | ChannelZ | that's probably what it's whining about |
| 10:03.55 | ChannelZ | I don't know why I typed those |
| 10:04.28 | freetown | ChannelZ, :-D, yeah the help returned alright |
| 10:04.49 | freetown | asterisk reports playing music with the m option |
| 10:04.53 | freetown | maybe it's too quiet? |
| 10:05.02 | freetown | is there a volume setting? :p |
| 10:05.21 | ChannelZ | have you use MOH before/have any songs even there? |
| 10:05.21 | freetown | man, i have to go before they lock me in |
| 10:05.39 | freetown | ChannelZ, just tested with softphone - works |
| 10:05.45 | ChannelZ | the music? |
| 10:05.47 | freetown | a bit on the quiet side |
| 10:05.51 | freetown | yes, music |
| 10:05.57 | ChannelZ | But you didn't hear the music on your cell? |
| 10:06.06 | freetown | yup |
| 10:06.08 | ChannelZ | something funky is happening on your ITSP link. |
| 10:06.16 | ChannelZ | I can't even fathom what |
| 10:06.40 | freetown | i better call up the ITSP then |
| 10:06.46 | *** join/#asterisk hrolf_ (~hrolf@unaffiliated/hrolf) |
| 10:06.59 | freetown | ChannelZ, thanks again for the time |
| 10:07.17 | ChannelZ | the music thing really makes no sense. You're not re-inviting so I don't know how they would even know you're making another call. |
| 10:07.47 | ChannelZ | sure. I'm really stumped. I'll be interested to hear the ultimate solution because I'm out of ideas |
| 10:08.22 | freetown | gotcha. see you in 14 hours |
| 10:08.30 | ChannelZ | :) |
| 10:09.32 | ChannelZ | I guess the final thing to look at is a complete SIP debug showing the traffic to/from your ITSP as well, for fun. But goodnight for now |
| 10:11.50 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
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| 10:21.54 | mechbangirc | is it possible to give caller an option "press 0 to leave a message" while caller is in queue on MOH? |
| 10:25.57 | kaldemar | mechbangirc: yes, define a context for the queue in queues.conf, it will enable single digit extensions in that context for queued callers. |
| 10:26.52 | mechbangirc | kaldemar: wow thanks dude. I never thought it would be this much easy. |
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| 10:40.28 | hrolf | Can I have something like this as a dialing exten? Like exten => _A2Q:[A-Za-z0-9]*:[A-Za-z0-9],1,AGI(...) ? |
| 10:40.54 | hrolf | So this way I'll be able to dial _A2Q:AnyText:AnyText ? |
| 10:42.19 | mandla | Help, http://pastebin.com/XK8pYaDC |
| 10:47.11 | kaldemar | hrolf: _.[:].[:]. is closest to that. |
| 10:47.35 | kaldemar | or _A2Q[:].[:]. |
| 10:48.22 | hrolf | kaldemar: What does '.' mean? |
| 10:50.46 | kaldemar | hrolf: one or more characters of anything. |
| 10:53.39 | hrolf | kaldemar: Is it a regex or it is specific to asterisk? |
| 10:55.11 | kaldemar | . is a single character in regex. this is asterisk-specific in patterns. |
| 10:55.27 | *** join/#asterisk irroot (~gregory@197.174.65.77) |
| 10:55.45 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 11:04.31 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 11:04.45 | IsUp | hello |
| 11:08.12 | defswork | any suggestions on chanspy dropping the listening party after a while ? |
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| 12:11.22 | Tim_Toady | i have an agi script (perl) that creates some sound files, plays them back and deletes them on exit, if the user hangs the call before the playback is completed script terminates and files are left on the disk. Is there a way to avoid it? |
| 12:12.28 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:14.43 | kaldemar | Tim_Toady: either set AGISIGHUP variable to "no" before the AGI execution or use DeadAGI. |
| 12:15.23 | Tim_Toady | deadagi is removed in asterisk10 right? so im trying to avoid it |
| 12:15.40 | Tim_Toady | i ll check the other, thx kaldemar |
| 12:18.33 | kaldemar | where did you come up with that? |
| 12:19.17 | kaldemar | DeadAGI is not removed in 10. |
| 12:21.08 | kaldemar | it was just deprecated in 1.8, but it still is available. the AGISIGHUP method is preferred though. |
| 12:22.33 | *** join/#asterisk irroot (~gregory@197.175.243.233) |
| 12:22.55 | Tim_Toady | ah... I had the impression it was removed. Thats what you get for doing stuff at 5 am :P |
| 12:34.44 | leifmadsen | you shouldn't really need to use DeadAGI though because I'm pretty sure AGI() just handles everything now |
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| 12:37.21 | kaldemar | yep, a DeadAGI call even prints the deprecation note. |
| 12:39.29 | dym | What is the exact definition of a "Span"? |
| 12:53.45 | Tim_Toady | AGISISGHUP does the job for now but im thinking a better approach would be to create a signal handler in the script and clean the files when recieving a SIGHUP |
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| 13:26.44 | ixyd_ | hi everyone, i would like to check if a queuemember is paused or unpaused in the dialplan...is there any way to get this information directly from the (realtime)backend, like a function or application? the only way i see is using a astdb or a custom devstate, but that wouldnt work in failover situations...any hints? :) |
| 13:29.37 | asteriskATmarmuD | hi guys. I am experiencing something strange. I got no DAHDI hardware installed and I am only using SIP phones. but I am using dahdi_dummy for timing purposes. any idea why I get the following warning? WARNING[10029]: channel.c:3740 ast_request: No channel type registered for 'DAHDI' |
| 13:31.21 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:32.10 | kaldemar | asteriskATmarmuD: you're trying to dial a DAHDI channel, and you don't have any. |
| 13:38.16 | asteriskATmarmuD | <PROTECTED> |
| 13:41.11 | kaldemar | asteriskATmarmuD: your dialplan is. somewhere you have "Dial(DAHDI/..." or some variable first inside a Dial command that begins with "DAHDI/". |
| 13:41.26 | kaldemar | asteriskATmarmuD: can you reproduce it? |
| 13:42.07 | asteriskATmarmuD | kaldemar: thx again. I will look into the dialplan again and search for DAHDI |
| 13:45.37 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 13:51.34 | irroot | ok guys be honnest who in the US is not haveing turkey for work |
| 13:54.26 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:54.46 | miztic | no turkey here |
| 13:57.33 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 13:58.46 | pabelanger | wild turkey and milk? |
| 13:58.49 | pabelanger | leifmadsen: ^ |
| 13:59.02 | leifmadsen | pabelanger: that's my favourite breakfast drink! |
| 14:01.22 | *** join/#asterisk cusco (~tralala@a79-168-174-232.cpe.netcabo.pt) |
| 14:01.23 | cusco | hi |
| 14:01.31 | cusco | I have a channel that I can't terminate |
| 14:01.39 | cusco | and cli is showing: [Nov 28 14:00:41] WARNING[24468]: app_meetme.c:3338 conf_run: Unable to write frame to channel SIP/150-00000389 |
| 14:01.42 | cusco | over and over |
| 14:02.05 | cusco | it was basically a MeetMe(1234,r,); |
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| 14:07.05 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:30.41 | *** join/#asterisk [Outcast] (~outcast@westford-nat.juniper.net) |
| 14:31.00 | [Outcast] | does asterisk have rtmp support? |
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| 14:32.34 | kaldemar | [Outcast]: no. |
| 14:32.42 | [Outcast] | :( |
| 14:32.50 | coppice | Freeswitch supports RTMP |
| 14:32.57 | [Outcast] | i know |
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| 14:40.58 | schmidts | coppice Freeswitch developer eat little children also :D |
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| 14:42.06 | coppice | I thought it was Belgians who did that. At least that's what they told the British public |
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| 14:45.06 | schmidts | coppice LOL wdoekes2 what can you say about this? |
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| 15:05.08 | as001 | Hi can you recommend me little box for Asterisk where I can put 1 fxo and 1 fxs modules. (Which can connect to 1 telephone line and 1 telephone device) ? |
| 15:05.37 | as001 | It should be conencted to internet too of course |
| 15:06.55 | [TK]D-Fender | Linksys SPA-3102 |
| 15:07.13 | as001 | ok Thanks [TK]D-Fender |
| 15:07.40 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:08.54 | MrTelephone | Is it normal for a channel to go ZOMBIE on transfer ? |
| 15:09.32 | MrTelephone | == Spawn extension (macro-stdexten, s, 5) exited non-zero on 'SIP/601-0000959a<ZOMBIE>' in macro 'stdexten' |
| 15:09.33 | MrTelephone | <PROTECTED> |
| 15:10.19 | stix | I guess it's normal. You'll probably see some renames too |
| 15:10.32 | schmidts | MrTelephone yes thats the way how it works |
| 15:11.48 | MrTelephone | ok just checking. I had a glitch where ${ARGV1} was being over written by something else in a macro. seen those ZOMBIE messages and was curious |
| 15:12.37 | schmidts | MrTelephone when you do a transfer you normally have 3 calls and one of them dies after the transfer is completed and this one is marked as zombie ;) |
| 15:12.56 | MrTelephone | ok. I just wondering why it woulnd't exit on 0 (no error)? |
| 15:13.07 | MrTelephone | since it's a normal operation :) |
| 15:13.28 | schmidts | non-zero doesnt means bad |
| 15:14.22 | MrTelephone | k |
| 15:16.27 | francisvgarcia | can anybody call me at sip:francisvgarcia.dyndns.org ? |
| 15:17.09 | MrTelephone | can you safely use nested macros? |
| 15:17.19 | dym | francisvgarcia: will you moan? :( |
| 15:17.26 | francisvgarcia | no |
| 15:17.27 | francisvgarcia | lol |
| 15:17.31 | francisvgarcia | It's just for testing |
| 15:17.34 | dym | then im out. |
| 15:17.35 | dym | (: |
| 15:17.35 | schmidts | ok then i will not call you :D |
| 15:17.39 | dym | hahaha |
| 15:17.42 | dym | o/ schmidts |
| 15:17.54 | schmidts | o/ dym |
| 15:18.21 | francisvgarcia | I want to deploy a click to dial in a web page |
| 15:18.45 | francisvgarcia | and I want to be sure that anybody from internet can call me |
| 15:19.41 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:20.22 | dym | I dont use the phone much |
| 15:20.26 | as001 | So Thanks [TK]D-Fender it is Asterisk what is working on that Cisco Small Business Pro SPA3102 ? I can configure it via ssh like I do with asterisk on clasic computer ? |
| 15:20.51 | ChannelZ | the 3102 has a web interface |
| 15:21.01 | dym | compiles asterisk on [TK]D-Fender's classic computer |
| 15:21.15 | [TK]D-Fender | it is just the gateway, not a full PS. Go slap a netbook or other embeded device with it |
| 15:21.58 | as001 | ok |
| 15:24.13 | *** join/#asterisk bmg505 (~leon@196-209-123-3.dynamic.isadsl.co.za) |
| 15:24.30 | as001 | is there any other device which is AsteriskPBX with one FXO, one FXS port and ethernet in some small box which can work as standalone without netbook ? |
| 15:24.52 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:24.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:26.09 | schmidts | as001 i know a small SOHO internet router which can do this, maybe you can take a look at the openRG project for devices they support |
| 15:26.23 | as001 | hmm ok |
| 15:27.11 | as001 | thanks schmidts |
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| 15:30.31 | [TK]D-Fender | as001, They are typically extremely limited in horsepower and storage and you might find yourself backed up into a corner depnding what you intend to get out of it. |
| 15:30.38 | *** part/#asterisk as001 (~uros@82.117.198.142) |
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| 15:42.08 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 15:52.20 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 15:52.39 | as001 | DId someone play with this device ipPBX02 ? It looks like that what I need |
| 15:52.49 | as001 | http://www.nicherons.com/ippbx02.html |
| 15:55.43 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:06.46 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 16:07.46 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 16:11.29 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 16:14.05 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 16:15.45 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:18.30 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 16:18.52 | [TK]D-Fender | as001, PBX software: Asterisk 1.4.21 |
| 16:18.55 | voipeng | for asterisknow 1.7 is the default cli password still maint and password? |
| 16:19.09 | as001 | yes old one I noticed that |
| 16:19.10 | [TK]D-Fender | Ancient, a security risk, and I'd want to be damn sure about what your upgrade path looks like |
| 16:19.38 | [TK]D-Fender | Which I'm betting is nearly non-existant |
| 16:20.26 | voipeng | nm got in i changed it already |
| 16:20.29 | as001 | I don't know I still dont have that device |
| 16:20.49 | [TK]D-Fender | as001, What are your real needs? |
| 16:21.45 | as001 | small always turn on device with linux and asterisk and 1 fxo 1fxs port and network adapter so I can connect telephone line and device and internet connection to it and control it with ssh |
| 16:22.17 | as001 | to make configuration by editing asterisk configuration files :) |
| 16:22.41 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 16:24.19 | as001 | like that one from link but with Asterisk 1.8 and Linux (not uClinux i dunno what is it...) |
| 16:24.56 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 16:25.03 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 16:25.31 | SeRi | as001: make your own netop system |
| 16:29.18 | as001 | I want to put that device in my home and to use legacy telephone device to call international calls via internet and to call local calls via present telephone line thanks to Asterisk dialplan... I don't think that netop can help me. |
| 16:29.25 | *** join/#asterisk timahvo1 (~rogue@197.178.196.200) |
| 16:30.11 | as001 | on the other hand I dont want to caputre my present computer to be just PBX which is always turn on in power... |
| 16:31.51 | [TK]D-Fender | Netbook = cheap & flxible |
| 16:32.19 | [TK]D-Fender | low power, built in battery backup, built in console. |
| 16:32.25 | *** join/#asterisk moy (~moy@216.172.42.74) |
| 16:32.27 | as001 | and why not in future to put that in as many homes I can and call all of them for free via internet... :) |
| 16:32.43 | as001 | ok I will check netbook Fender |
| 16:32.44 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:33.19 | as001 | but there must be some device capable for that like that above with ancient asterisk 1.4.21 |
| 16:34.38 | *** join/#asterisk chasing`Sol (~cS@41.206.151.27) |
| 16:37.36 | as001 | thanks for help Fender. ChannelZ i remember your help from week ago.. |
| 16:39.32 | *** join/#asterisk Stratisphere (~Stratisph@unaffiliated/stratisphere) |
| 16:40.59 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
| 16:43.30 | SeRi | as001: of course a netop can do that... |
| 16:43.48 | SeRi | You will need to buy a FXS/FXO card and thats it. |
| 16:45.07 | SeRi | as001: I have no idea where you got that a netop can not do that. is fully doable. I am building one all ready. |
| 16:46.01 | Stratisphere | anyone know off the top of their head what the dial string is for the sangoma a500? currently I have WOOMERA/g0/$OUTNUM$ |
| 16:46.46 | as001 | ok I just looked a quickly netop and think it is remotely manager system far away from my place impossible to connect to my phone line.. I will check again thanks for help |
| 16:46.50 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 16:46.59 | gandhijee | hey are the poly IP4000 speaker phones PoE ready or do they need some funky adapter |
| 16:47.30 | _Corey_ | gandhijee: The IP4000s require the funky adapter |
| 16:47.54 | gandhijee | thanks |
| 16:48.18 | _Corey_ | if you're buying new, go ip6000 |
| 16:48.30 | SeRi | wow really that guy is clueles... lol |
| 16:48.30 | gandhijee | how much do those run? |
| 16:48.50 | _Corey_ | They're the successor to the ip4000, so about the same |
| 16:50.15 | gandhijee | i am gonna go out on a limb and guess that the polycom speaker phones are the only ones worth getting as well |
| 16:50.30 | _Corey_ | Yeah |
| 16:50.43 | _Corey_ | Cisco, etc. slaps their logo on them |
| 16:51.31 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v029-126.mobile.uci.edu) |
| 16:52.07 | Naikrovek | the conf phones, yeah, probably |
| 16:52.24 | Naikrovek | unless you take them apart and do a circuit board comparison that's just a guess but the design is certainly polycom |
| 16:56.17 | SeRi | p3nguin: you avail? |
| 16:56.33 | p3nguin | dun dun dunnnn |
| 16:56.39 | SeRi | lol |
| 16:56.58 | SeRi | well they work. boy do they do. |
| 16:57.03 | SeRi | lol |
| 16:57.13 | p3nguin | Awesome! |
| 16:58.38 | p3nguin | Just remember, take one about 20 minutes before you are ready for bed. |
| 16:58.39 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:59.03 | SeRi | p3nguin: Yea I lean that last night I tought it would take longer but it was pretty fast |
| 16:59.31 | p3nguin | Good thing you didn't get 5mg or 6mg. |
| 17:00.18 | SeRi | lol |
| 17:00.44 | p3nguin | Class Policy Sent Rate Dropped Overlimit Backlog |
| 17:00.45 | p3nguin | root shaper 83374538 0 17 0 |
| 17:00.57 | p3nguin | There was a lost call already, but it wasn't the same as before. |
| 17:01.05 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 17:01.13 | p3nguin | This time, the remote phone just disappeared and there was no SIP warning. |
| 17:01.14 | SeRi | what happen? |
| 17:01.20 | SeRi | o shit |
| 17:01.35 | p3nguin | Since it was different, I don't know if it was the same problem or a new one. |
| 17:01.48 | SeRi | mhhhh |
| 17:02.12 | p3nguin | It was 1.000000m duration. Could be a coincidence. |
| 17:02.15 | SeRi | no logs on your firewall? |
| 17:02.22 | SeRi | I see |
| 17:02.29 | *** part/#asterisk Stratisphere (~Stratisph@unaffiliated/stratisphere) |
| 17:02.47 | *** join/#asterisk outtolunc (~outtolunc@adsl-66-218-53-172.dslextreme.com) |
| 17:04.46 | SeRi | p3nguin: sound like it... |
| 17:08.49 | SeRi | its weird though..... |
| 17:09.28 | p3nguin | http://pastebin.com/wWtnnaQ1 |
| 17:09.32 | SeRi | brb breakfast is ready... :) french toast! |
| 17:10.17 | p3nguin | It's approaching lunch time. |
| 17:10.49 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 17:11.07 | p3nguin | I think I'll have roast beef and nacho cheese sauce on Texas toast. |
| 17:11.10 | d_preston215 | Is there a way to disable a person leaving a voicemail message from marking the message as urgent? |
| 17:11.30 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 17:11.38 | [TK]D-Fender | d_preston215, don't give then the review option on the box |
| 17:12.53 | *** join/#asterisk chasing`Sol (~cS@41.206.150.61) |
| 17:12.55 | d_preston215 | Thanks. |
| 17:14.18 | p3nguin | That's going to be for the mail box, though, not a single caller. |
| 17:18.32 | SeRi | p3nguin: That sounds good lol |
| 17:19.10 | SeRi | p3nguin: I see your over limits but that should not cause what happen earlier... I think that was a coinsidence |
| 17:19.28 | p3nguin | I'm trying to find out what causes root/shaper to have overlimits. |
| 17:19.49 | p3nguin | I thought everything not matched by one of my three classes should end up in default. |
| 17:20.21 | p3nguin | I'm not yet a vyatta expert, so I don't know how it happens. |
| 17:20.56 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 17:22.49 | SeRi | p3nguin: https://calomel.org/pf_hfsc.html |
| 17:22.59 | SeRi | not sure if that would help. |
| 17:23.05 | SeRi | I have qlimits |
| 17:23.11 | SeRi | you have "limits" |
| 17:23.59 | p3nguin | With multiple calls at once, I didn't have any overlimits: http://pastebin.com/2Np6gGjN |
| 17:24.37 | SeRi | strange.... |
| 17:28.32 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:32.05 | SeRi | p3nguin: was it pstn call? |
| 17:33.24 | p3nguin | The one that was lost earlier was, yes. It was the same scenario as before: remote phone A making a call though asterisk to voipms (to the PSTN). |
| 17:34.01 | *** part/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 17:34.11 | voipeng | in asterisknow after I ran all updates I did not see ngrep installed, is there another application used for this function or do i need to download and install it on the server? |
| 17:35.39 | [TK]D-Fender | yum <- like everything else |
| 17:36.02 | voipeng | yea yum install ngrep does not work |
| 17:36.03 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 17:36.14 | voipeng | wget the install and run the rpm? |
| 17:36.25 | p3nguin | You could. |
| 17:36.31 | SeRi | damn |
| 17:36.40 | p3nguin | But there were no sip_reg_timeout messages when it happened this time, so I'm inclined to think it was a different problem. |
| 17:36.55 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-18.nwrknj.fios.verizon.net) |
| 17:37.18 | Dovid | is there any way to have asterisk use the same call-id on both legs of a call? |
| 17:37.24 | p3nguin | ngrep is in the epel repo. YOu may have to enable it. |
| 17:37.30 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:37.34 | *** join/#asterisk carloimperia (~carloimpe@2.44.6.170) |
| 17:39.10 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.239.171.dsl.dyn.forthnet.gr) |
| 17:40.09 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:40.12 | wcselby | o/ |
| 17:41.09 | p3nguin | Salami |
| 17:42.30 | wcselby | bless you |
| 17:42.51 | voipeng | p3nguin: did i want epel 4,5,or 6? |
| 17:42.53 | SeRi | waz up wcselby... enjoying this clod ass weather? |
| 17:43.26 | wcselby | heh |
| 17:43.26 | wcselby | yeah |
| 17:43.30 | SeRi | lol |
| 17:43.49 | wcselby | it's like 50 degrees, not too bad :) |
| 17:44.16 | SeRi | this morning was like 35 fuck that |
| 17:44.22 | wcselby | lol yeah |
| 17:44.32 | p3nguin | voipeng: Match your OS version. I have CentOS 5 and use epel-5. |
| 17:44.39 | voipeng | thanks |
| 17:44.50 | wcselby | I thought I was going to have to scrape the ice, but then I realized I could just pull back two feet and let it sit in the sun for two minutes :) |
| 17:45.27 | SeRi | lol |
| 17:45.29 | SeRi | brb |
| 17:46.06 | p3nguin | CentOS 5.5 to be more specific. http://pastebin.com/fKFKgLzr |
| 17:46.32 | SeRi | yay! is fixed |
| 17:46.34 | SeRi | lol |
| 17:46.38 | SeRi | brb |
| 17:47.06 | p3nguin | 39 degrees here right now. |
| 17:47.19 | p3nguin | That's 4 C. |
| 17:49.37 | *** join/#asterisk AdamN (~AdamN@63.230.70.254) |
| 17:49.46 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 17:49.55 | voipeng | i am trying to setup a trunk to our production network. I am looking at a ngrep from the asterisknow server and I do not see it trying to register... |
| 17:50.22 | p3nguin | Use the built-in sip debug. |
| 17:50.38 | [TK]D-Fender | voipeng, * CIL SIP DEBUG. |
| 17:50.42 | [TK]D-Fender | CLI |
| 17:50.52 | p3nguin | sip set debug on |
| 17:51.04 | voipeng | [root@wlcpbx01 ~]# * cli sip debug |
| 17:51.04 | voipeng | -bash: anaconda-ks.cfg: command not found |
| 17:51.52 | navaismo | ~cli |
| 17:51.52 | infobot | methinks cli is a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
| 17:52.03 | navaismo | ? |
| 17:52.13 | AdamN | has anyone else stopped recieving new voicemail notifications in the last asterisk update? |
| 17:52.43 | [TK]D-Fender | voipeng, asterisk -rvvvvvvvvvvvv |
| 17:52.45 | SeRi | bbl guys. wife left me a "list" of stuff to go buy. |
| 17:52.50 | SeRi | cya guys! |
| 17:53.34 | pabelanger | navaismo: possible, I know there was some recent fixes for voicemail and notifications. You should try 1.8.8.0-rc3 |
| 17:54.08 | voipeng | D-Fender: wlcpbx01*CLI> sip debug |
| 17:54.08 | voipeng | No such command 'sip debug' (type 'core show help sip debug' for other possible commands) |
| 17:54.08 | voipeng | wlcpbx01*CLI> sip debug enable |
| 17:54.36 | pabelanger | ~collectdebug |
| 17:54.36 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:54.38 | pabelanger | voipeng: ^ |
| 17:54.40 | [TK]D-Fender | <p3nguin> sip set debug on |
| 17:54.42 | [TK]D-Fender | ^^^^66 |
| 17:55.08 | voipeng | wlcpbx01*CLI> sip set debug on |
| 17:55.08 | voipeng | No such command 'sip set debug on' (type 'core show help sip set' for other possible commands) |
| 17:55.14 | *** join/#asterisk pdtpatrick1 (~pdtpdt@12.249.4.226) |
| 17:55.15 | navaismo | AdamN: i think the response of pabelanger is for you |
| 17:55.20 | [TK]D-Fender | voipeng, What version are you running? |
| 17:55.48 | p3nguin | Nothing modern. |
| 17:55.50 | pabelanger | navaismo: mybad |
| 17:55.57 | pdtpatrick1 | Question .. how does one avoid agents and/or queues going invalid? Right now they are set to dynamic which i believe is the problem. Any suggestions/links would be greatly appreciated |
| 17:56.01 | voipeng | i check the version from the cli i take it? |
| 17:56.28 | [TK]D-Fender | voipeng, when you connected to CLI it should have told you |
| 17:56.39 | p3nguin | If "sip set debug on" does not work, you're running an old software and should try "sip set debug" instead. |
| 17:56.40 | voipeng | translation uptime version warranty |
| 17:56.41 | voipeng | wlcpbx01*CLI> core show version |
| 17:56.41 | voipeng | Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-24 20:43:18 UTC |
| 17:56.57 | voipeng | wlcpbx01*CLI> sip set debug |
| 17:56.57 | voipeng | No such command 'sip set debug' (type 'core show help sip set' for other possible commands) |
| 17:56.57 | voipeng | wlcpbx01*CLI> |
| 17:57.04 | voipeng | I just installed the lastest asterisknow |
| 17:57.10 | voipeng | and ran yum updates after installation |
| 17:57.14 | p3nguin | Okay, so you don't have any SIP channel. Lovely. |
| 17:57.27 | p3nguin | That would be why your shit does not work. |
| 17:57.34 | voipeng | nice |
| 17:57.41 | [TK]D-Fender | voipeng, "sip show peers" <- |
| 17:58.12 | d_preston215 | Its bad to use leastrecent ringstrategy with extensions with call waiting on them, right? |
| 17:58.56 | voipeng | wlcpbx01*CLI> sip show peers |
| 17:58.57 | voipeng | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 17:59.04 | [TK]D-Fender | voipeng, chan_sip isn't even loaded |
| 17:59.14 | [TK]D-Fender | voipeng, You've fubar'd your configs |
| 17:59.32 | voipeng | i havent attempted to configure anything but the trunk yet, not sure what I could have screwed up |
| 18:00.03 | [TK]D-Fender | voipeng, pastebin "ls -la /etc/asterisk" and your sip.conf and every INCLUDE-d file |
| 18:00.20 | [TK]D-Fender | ~pb |
| 18:00.20 | 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. |
| 18:00.21 | [TK]D-Fender | ^^^ |
| 18:02.43 | voipeng | gotcha im familar with pastebin now |
| 18:02.48 | voipeng | and i did the ls -la |
| 18:03.00 | voipeng | so you want sip.conf, what is include-d file? |
| 18:03.12 | [TK]D-Fender | if you use the INCLUDE directive to include other files |
| 18:03.13 | voipeng | every file in that directory? |
| 18:03.48 | pdtpatrick1 | QUestion .. does anyone know a good python API to talk to asterisk ? |
| 18:06.24 | pabelanger | pdtpatrick1: starpy |
| 18:06.45 | [TK]D-Fender | trunk not passing on proper progress |
| 18:06.50 | [TK]D-Fender | oops. |
| 18:07.16 | voipeng | dfender, im not sure what other files you needed? here is the sip.conf http://pastebin.com/WpfwzZUy |
| 18:07.57 | [TK]D-Fender | #include sip_custom.conf <--- INCLUDES |
| 18:08.08 | voipeng | ahhh ok |
| 18:08.09 | voipeng | sorry |
| 18:08.15 | [TK]D-Fender | voipeng, and pastebin your modules.conf while you're at it |
| 18:08.29 | voipeng | d-fender: ok |
| 18:08.47 | [TK]D-Fender | voipeng, pastebin "ls -la /etc/asterisk" <----- |
| 18:08.59 | [TK]D-Fender | mask your PW's BTW |
| 18:09.13 | voipeng | shit there was a password in there? |
| 18:09.31 | voipeng | or your saying in the next files |
| 18:10.11 | [TK]D-Fender | nest files |
| 18:10.13 | voipeng | http://pastebin.com/CMVYQ6e7 - ls -la |
| 18:10.14 | [TK]D-Fender | next |
| 18:11.03 | [TK]D-Fender | -rw-rw-r-- 1 asterisk asterisk 619 Nov 28 13:40 sip_general_additional.conf |
| 18:11.19 | [TK]D-Fender | -rw-rw-r-- 1 asterisk asterisk 418 Nov 28 13:40 sip_registrations.conf |
| 18:11.26 | voipeng | doing modules.conf now |
| 18:11.27 | [TK]D-Fender | -rw-rw-r-- 1 asterisk asterisk 614 Nov 28 13:40 sip_additional.conf |
| 18:11.31 | voipeng | what was the file that has the password? |
| 18:11.32 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
| 18:11.44 | [TK]D-Fender | those 3, along with modules.conf. make sure to mask all PW's |
| 18:12.17 | [TK]D-Fender | voipeng, And actually.. just do "module load chan_sip.so" at * CLI |
| 18:12.25 | [TK]D-Fender | lets see if you can load it manually |
| 18:12.34 | voipeng | http://pastebin.com/egjPYBM7 for module |
| 18:12.39 | voipeng | k ill try that before i get the other 3 files |
| 18:13.31 | voipeng | done - http://pastebin.com/V9awCPCs |
| 18:14.05 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 18:15.11 | [TK]D-Fender | ok, it loaded |
| 18:15.17 | [TK]D-Fender | forget the other PB now |
| 18:15.21 | voipeng | ok |
| 18:15.26 | [TK]D-Fender | I am wondering why it didn't load previously |
| 18:15.35 | [TK]D-Fender | but I'll leave that for now |
| 18:15.35 | voipeng | i just did the yum updates this morning? |
| 18:15.39 | p3nguin | FreePBX support? |
| 18:15.43 | voipeng | didnt reboot system since |
| 18:15.44 | wcselby | anyone here actually use the windows backup tool to backup their windows servers? |
| 18:15.49 | [TK]D-Fender | This of course explains your previous failures. No SIP at all |
| 18:15.55 | voipeng | hah |
| 18:16.07 | p3nguin | Nah, I guess FreePBX doesn't control that part. |
| 18:16.11 | [TK]D-Fender | voipeng, Ok, go try stuff now |
| 18:16.30 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 18:16.42 | voipeng | anyway i can force the trunk to try and re-register? |
| 18:16.55 | voipeng | or do i save changes in the webgui and then apply changes |
| 18:16.57 | blizzow | I get lots of complaints in my call center about Zoiper biz. Does anyone here have a recommendation for a good windows SIP client? |
| 18:18.59 | [TK]D-Fender | voipeng, It should have jsut registered. "sip show registry" "sip show peers" and when in doubt do "sip set debug on" and "sip reload" and watch the new attempt |
| 18:19.17 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 18:21.27 | voipeng | d-fender: ran those commands : http://pastebin.com/g0h2gkCb |
| 18:21.56 | voipeng | one peer is registration by ip only the other is by username |
| 18:23.08 | [TK]D-Fender | voipeng, So you've double entered your trunks, or did a split suer/peer in a probably wrong way |
| 18:23.30 | [TK]D-Fender | voipeng, You also do not appear to have told it to register in the first palce |
| 18:23.33 | [TK]D-Fender | place* |
| 18:23.44 | voipeng | k np, i can blow them both out and do it correctly |
| 18:23.50 | voipeng | whats the right way :) |
| 18:24.00 | [TK]D-Fender | I don't know what you hav to configure. |
| 18:24.13 | voipeng | trunk from this asterisknow to our production server |
| 18:24.15 | [TK]D-Fender | maybe they need 2 sections, maybe they don't I don't know what you put in, or why |
| 18:24.29 | voipeng | i can do by ip or by username and pass.. |
| 18:24.33 | [TK]D-Fender | voipeng, And what is this "productions server"? |
| 18:24.47 | voipeng | asterisk 1.4 something and voiceaxis |
| 18:25.02 | [TK]D-Fender | Oh yes.. that mess... |
| 18:25.06 | voipeng | trying to offload applications that dont work there |
| 18:25.10 | [TK]D-Fender | single peer. |
| 18:25.10 | voipeng | here if i can get it to work |
| 18:25.11 | voipeng | yep |
| 18:25.17 | voipeng | at the moment yes |
| 18:25.32 | [TK]D-Fender | No, that is what you need |
| 18:25.35 | voipeng | trying to test a trunk from my context my voip phone is on to the asterisknow box |
| 18:25.39 | voipeng | ok |
| 18:25.42 | [TK]D-Fender | you sholdn't need a "user" section of a trunk, just the top part of one. |
| 18:26.00 | [TK]D-Fender | Please continue in #freePBX as we have long left the scope of this channel |
| 18:26.00 | voipeng | ? so do it ip based? |
| 18:26.12 | voipeng | heh thanks.. |
| 18:28.50 | *** join/#asterisk francisvgarcia (~francis.g@190.80.239.124) |
| 18:31.38 | p3nguin | I'd tell you how to do it in Asterisk, but that won't do you any good once you press the big orange Apply button in FreePBX. |
| 18:31.50 | voipeng | hah |
| 18:32.05 | voipeng | cant i manually do it your way through the cli? |
| 18:32.07 | voipeng | or ssh |
| 18:32.21 | voipeng | havent gotten a response from the freepbx channel yet |
| 18:32.46 | [TK]D-Fender | You asked 1 MINUTE ago |
| 18:32.51 | voipeng | lol |
| 18:32.55 | voipeng | sorry |
| 18:46.02 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 18:46.06 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 18:47.59 | voipeng | d-fender: posted the images you asked for |
| 18:57.18 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:09.27 | *** part/#asterisk ulogic (~root@ool-4a59e7fc.dyn.optonline.net) |
| 19:16.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:16.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:17.20 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 19:20.07 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:23.50 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 19:27.27 | Dovid | anyone here use sipp |
| 19:29.06 | *** join/#asterisk kikohnl (~kotis@ext-dip-166.hnl.cdsinc.com) |
| 19:36.02 | leifmadsen | yes |
| 19:41.07 | p3nguin | You always get the easy questions. |
| 19:43.21 | jkroon | p3nguin, that's not an easy question. |
| 19:43.27 | p3nguin | (1327.27) <Dovid> anyone here use sipp |
| 19:43.29 | p3nguin | (1336.02) <@leifmadsen> yes |
| 19:43.35 | p3nguin | Very easy! |
| 19:43.36 | leifmadsen | seemed straight forward to me :) |
| 19:43.55 | jkroon | it's dead obvious. |
| 19:45.04 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:52.07 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:00.30 | Dovid | leifmadsen: Sorry. was busy fighting with sipp. i think i figued it out |
| 20:00.36 | leifmadsen | ok |
| 20:00.45 | leifmadsen | don't be sorry, I already answered your question |
| 20:00.50 | Dovid | hehe |
| 20:00.53 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 20:00.55 | leifmadsen | wasn't waiting with baited breath |
| 20:05.24 | hardwire | what does that even mean? |
| 20:05.29 | hardwire | is it like a vampire thing? |
| 20:05.37 | hardwire | I can't imagine anything being attracted via breath. |
| 20:07.15 | hardwire | http://www.worldwidewords.org/qa/qa-bai1.htm |
| 20:07.17 | hardwire | well.. now I know. |
| 20:07.44 | leifmadsen | hardwire: aye, although I used the wrong "baited" |
| 20:07.57 | leifmadsen | hardwire: for anyone who is too lazy to look, it means, "waiting with anticipation" |
| 20:08.07 | hardwire | leifmadsen: apparently you did or didn't.. depending on who cares :) |
| 20:08.19 | leifmadsen | hardwire: probably depends if you're British or French :) |
| 20:08.33 | hardwire | or shakespeare. |
| 20:11.38 | *** join/#asterisk Ast001 (~uros@cable-89-216-173-83.dynamic.sbb.rs) |
| 20:12.41 | Ast001 | Hello again. SeRi you were right about nettop I wonder which model do you use and how can I put digiurm card inside. It looks like small device. |
| 20:12.57 | Ast001 | *digium |
| 20:14.11 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 20:15.20 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 20:19.35 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 20:19.48 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 20:20.05 | kuku | I have canreinvite=no everywhere, and it still reinvites... any clues ? |
| 20:21.15 | _Corey_ | kuku: It's called 'directmedia' now |
| 20:21.30 | [Outcast] | kuku, you using nat and what version? |
| 20:24.01 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 20:25.21 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-135-174.pools.arcor-ip.net) |
| 20:28.50 | navaismo | anyone has used the syslog option with grandstream phones? |
| 20:34.59 | kuku | [Outcast]: using nat on some - yes |
| 20:36.05 | SeRi | Ast001: some netops have pci,pcie, and or pcix, options |
| 20:36.19 | SeRi | is really up to how would you like to build it. |
| 20:37.09 | Ast001 | I need 1 fxo and 1 fxs port card I already have such card but I dont know which model of netops to search for |
| 20:37.24 | kuku | [Outcast]: 1.8.2.3 |
| 20:38.25 | Ast001 | so I need netop with pci slot |
| 20:40.05 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 20:40.18 | wcselby | hello again |
| 20:42.04 | [Outcast] | kuku: read the first little bit here: http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
| 20:46.15 | p3nguin | seri: I made some changes to the shaper: http://pastebin.com/nJ6b5c52 |
| 20:50.24 | tzanger | I'm playing around with sip auto-registration (autocreatepeer=yes) -- I dump them into a dialplan that I do my own authentication (PIN) in before I allow a call to be routed |
| 20:50.58 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:51.08 | tzanger | the question is about overriding/ignoring the phone-supplied username. Is there a way to override or replace what the phone gives me as a username, or a way to create a guaranteed-unique peer name? |
| 20:51.31 | tzanger | right now if phone A registers as Phone1 and phone B also registers as Phone1 I am concerned about mis-routing calls |
| 20:54.07 | SeRi | Ast001: Yes |
| 20:54.12 | SeRi | p3nguin: looking at it now |
| 20:55.47 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 20:56.10 | SeRi | p3nguin: I see you distributed the over all % better. also you have a class now set? |
| 20:59.08 | p3nguin | "also you have a class now set?" I do not understand this question. |
| 21:00.10 | p3nguin | I have the same three explicit classes and the default class that I have always had. The only thing I've changed today is the bandwidth and ceiling on each of the four classes. |
| 21:03.17 | SeRi | I didnt notice the class.... Yes the % looks better distribute now. |
| 21:04.19 | *** part/#asterisk Ast001 (~uros@cable-89-216-173-83.dynamic.sbb.rs) |
| 21:07.11 | p3nguin | I don't know that the changes will have any effect on the dropped calls, though. |
| 21:07.15 | *** join/#asterisk master_of_master (~master_of@p57B5519E.dip.t-dialin.net) |
| 21:08.21 | SeRi | well the celing might make a difference but I doubt your drop calls are caused by that. |
| 21:11.29 | WIMPy | Why do you set a ceiling? |
| 21:23.09 | p3nguin | Why not? |
| 21:26.13 | p3nguin | wimpy: I did want to let you know that the vyatta traffic shaper does give unused bandwidth to another class when needed. I know you said you wanted that functionality from a shaper. The bandwidth value is the guaranteed allocation and the ceiling is the limit on what can be used when other classes are not using their allocated bandwidth. |
| 21:26.48 | p3nguin | So if I set bandwidth of 10% and ceiling of 100%, and no other classes are using up their bandwidth, this would give all 100% to this class. |
| 21:26.56 | [TK]D-Fender | checkout time, BBIAB |
| 21:27.48 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hawqurajzsytgtcc) |
| 21:29.00 | WIMPy | That's exactely what TC does. |
| 21:29.43 | p3nguin | Vyatta is using TC in the back-end. |
| 21:30.11 | p3nguin | You just simply don't touch tc directly, just like you don't touch iptables directly. |
| 21:36.54 | p3nguin | More screwed up calls. I'm disabling the shaper completely. |
| 21:39.25 | SeRi | p3nguin: man that sucks that you are having those issues. |
| 21:40.30 | p3nguin | I'd bet the shaper is responsible for this one. The recording is VERY choppy and the people cannot hear each other. |
| 21:41.31 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 21:42.52 | p3nguin | "Thuh yuh fuh cuhluh Suhluh huhputuh. Ih yuh nuh duh extuh uh duh puh yuh uh cuhluh, pluh duh uh nuh..." |
| 21:42.56 | p3nguin | NO GOOD! |
| 21:45.01 | SeRi | not good at all :( |
| 21:45.14 | p3nguin | I've also noticed some jabber socket read errors during the call which is not working right. Maybe it's the modem or the interface on the router. |
| 21:45.20 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hawqurajzsytgtcc) |
| 21:45.35 | p3nguin | Hey, there goes a guy from Digium! |
| 21:45.37 | p3nguin | snickers |
| 21:45.43 | SeRi | lmao |
| 21:46.14 | p3nguin | I noticed that the Cisco guy hasn't been around during business hours today. |
| 21:46.26 | p3nguin | I think he left at 0851. |
| 21:47.47 | SeRi | lol |
| 21:50.22 | p3nguin | With a bandwidth of 24% and ceiling of 48% on default, perhaps that's why the stuff was really goofy on that last call. I think maybe all my packets are not matching the classes. |
| 21:52.09 | p3nguin | 24% of 2Mbit should be enough for calls to work, though. |
| 21:52.58 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-huehabtyugykoevd) |
| 21:54.37 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:57.32 | *** join/#asterisk happylife (~happylife@212.92.145.7) |
| 21:59.35 | SeRi | p3nguin: I am sure. |
| 21:59.46 | p3nguin | What would be the result of giving four classes each 50% bandwidth? |
| 22:00.18 | SeRi | I have mine set at real time top 15% and lowest 10% with a qlimit of 500 |
| 22:00.41 | p3nguin | I really want to give priority to RTP, so I know I want to give it as much bandwidth as I can spare. |
| 22:00.59 | p3nguin | I know I want SIP to stop dropping, so I want to give it as much as possible, too. |
| 22:01.16 | p3nguin | And then there's IAX2, which also needs as much as possible. |
| 22:01.27 | p3nguin | Everything else can suck on a twinkie. |
| 22:01.44 | SeRi | lol |
| 22:02.53 | SeRi | fuck |
| 22:02.58 | SeRi | I deleted a rule by accident |
| 22:02.59 | SeRi | damn |
| 22:03.30 | p3nguin | I'm averaging 1.92 Mbits/sec upstream on that system. |
| 22:04.13 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:04.32 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:04.36 | p3nguin | 1971 kbits/sec |
| 22:04.54 | p3nguin | And I set the bandwidth at 2000. |
| 22:05.45 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:07.11 | *** join/#asterisk kresp0 (~kresp0@178.200.217.87.dynamic.jazztel.es) |
| 22:07.23 | s[X] | hey p3nguin |
| 22:08.14 | *** join/#asterisk dmz (~dmz@67.216.138.246.pool.hargray.net) |
| 22:08.23 | SeRi | p3nguin: have you drop calls since you deleted the rules? |
| 22:08.29 | SeRi | or disable shaping? |
| 22:08.37 | SeRi | brb phone |
| 22:08.45 | s[X] | hey SRi |
| 22:08.46 | p3nguin | Not yet... but there are no calls. |
| 22:08.48 | s[X] | SeRi* |
| 22:11.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:14.58 | p3nguin | How can I find out of some of my SIP traffic going out does not have a source port of 5060? |
| 22:15.20 | p3nguin | I think some of the traffic isn't matching the policy class. |
| 22:18.43 | blizzow | Does anyone here know how to configure the Cisco IP communicator soft phone to work with asterisk over SIP? |
| 22:18.54 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 22:18.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:20.39 | SeRi | p3nguin: that depends on your nat translation |
| 22:21.00 | SeRi | some nat technologies re write the source port |
| 22:21.11 | SeRi | This is the case in pfsense |
| 22:21.36 | SeRi | so I have to set the out bound natting to static |
| 22:23.48 | SeRi | p3nguin: look at the types here: http://en.wikipedia.org/wiki/Network_address_translation |
| 22:24.58 | SeRi | p3nguin: here is the pfsense type for reference and to see hwy I have to set Out Bount NAT to static: http://doc.pfsense.org/index.php/Static_Port |
| 22:28.35 | SeRi | p3nguin: and I think you are in the same boat as me with symmetric nating. |
| 22:29.18 | p3nguin | I don't even understand that. |
| 22:29.50 | p3nguin | iptables doesn't do any magic as far as I can tell. If the host sends from port 5060, it'll go out 5060. |
| 22:30.41 | SeRi | Its not iptables. its nat and iptables is not doing your nat. or is it? |
| 22:31.48 | p3nguin | It is. |
| 22:33.40 | *** join/#asterisk dmz (~dmz@67.216.138.246.pool.hargray.net) |
| 22:33.56 | SeRi | You need to find out if vyatta is doing anything to the outbound destination port. |
| 22:34.12 | SeRi | can you set a ws after the router? |
| 22:34.36 | p3nguin | http://pastebin.com/jmdyTp8v |
| 22:34.46 | p3nguin | no |
| 22:35.55 | SeRi | p3nguin: do you have access to your states table? that would tell you what ports are been used as destination |
| 22:37.56 | SeRi | p3nguin: ok I see your iptables setup. here is an example of a good state.... 10.30.2.53:5060 -> WANIP:5060 -> 204.11.192.23:5060 |
| 22:39.32 | p3nguin | Pre-NAT src Pre-NAT dst Post-NAT src Post-NAT dst |
| 22:39.32 | SeRi | bad state: 10.30.2.53:5060 -> WANIP:58560 -> 204.11.192.23:5060 (Source Port re writing) |
| 22:39.35 | p3nguin | 192.168.192.242:5060 64.154.41.150:5060 75.123.123.123:5060 64.154.41.150:5060 |
| 22:40.43 | SeRi | p3nguin: well shit you are good. well knowing that iptables is doing your nat you shouldnt have any issues on that side |
| 22:41.28 | SeRi | p3nguin: so that answers your wuestion |
| 22:41.40 | p3nguin | No, actually it doesn't. |
| 22:41.46 | p3nguin | It answered *your* question. |
| 22:41.55 | SeRi | [16:14:58] < p3nguin> | How can I find out of some of my SIP traffic going out does not have a source port of 5060? |
| 22:42.02 | p3nguin | My question was whether or not all my SIP traffic always goes out 5060 or not. |
| 22:42.05 | SeRi | ^^ |
| 22:42.22 | SeRi | does not seem like that to me :/ |
| 22:42.53 | p3nguin | The question you just quoted and the reiteration are the same. |
| 22:43.37 | SeRi | If understood your question your asking how can you see if your traffic going out dot have the source port of 5060 |
| 22:43.50 | SeRi | s/dot/does/ |
| 22:44.21 | SeRi | the answer is "nat states" |
| 22:45.10 | p3nguin | no |
| 22:45.17 | p3nguin | I'm not talking about NAT at all. |
| 22:45.22 | p3nguin | I'm talking about SIP traffic. |
| 22:45.33 | SeRi | ok I see. |
| 22:46.05 | SeRi | other than using ws after the firewall or using pc on the firewall it would be hard to know.... |
| 22:46.29 | p3nguin | I can tshark the WAN interface. |
| 22:46.36 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 22:46.46 | *** join/#asterisk _-Jon-_ (~jon@74.198.87.94) |
| 22:46.53 | _-Jon-_ | Evening all |
| 22:47.03 | SeRi | perfect :) than thats the only idea I can think of to catch sip traffic and sport |
| 22:47.24 | SeRi | just listen for udp sip sessions or such... |
| 22:47.26 | p3nguin | But then we're back to where I started: how can I be sure that SIP traffic isn't using another port? |
| 22:48.04 | _-Jon-_ | This might be a silly question, but is there a simple way of blacklisting a bunch of numbers? |
| 22:48.16 | SeRi | well after it leaves the wan interface your catching it with tshark so that should tell you what sport it has or no? |
| 22:48.38 | leifmadsen | _-Jon-_: yes, just match them in the dialplan and hangup() |
| 22:48.59 | SeRi | p3nguin: one sec |
| 22:49.16 | leifmadsen | https://wiki.asterisk.org/wiki/display/AST/Function_BLACKLIST |
| 22:49.21 | leifmadsen | _-Jon-_: ^^^ |
| 22:49.46 | _-Jon-_ | Sweet, that is exactly what I need :) |
| 22:49.57 | leifmadsen | odd how looking up documentation is useful :) |
| 22:50.12 | _-Jon-_ | lol |
| 22:50.12 | p3nguin | seri: Let me see if I can be more clear. How will I be able to filter ONLY SIP TRAFFIC by something other than the port? If the port is NOT 5060, and if I do not know the port, how will I find it? |
| 22:50.30 | _-Jon-_ | leifmadsen, why are you here then? :) |
| 22:51.41 | p3nguin | I think a more appropriate question would be: Why are you here, when the documentation tells you what you wanted to know? |
| 22:52.02 | Nugget | he heard there were muffins |
| 22:52.09 | _-Jon-_ | And I like muffins |
| 22:52.16 | p3nguin | I don't blame you for that. |
| 22:52.20 | _-Jon-_ | :P |
| 22:52.34 | s[X] | ffs i really want a damm muffin now |
| 22:52.39 | _-Jon-_ | And I also like the attitude I get when asking simple questions |
| 22:52.39 | _-Jon-_ | :D |
| 22:55.06 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 23:01.02 | kresp0 | wtf! look: |
| 23:01.10 | kresp0 | chi*CLI> sip show peers |
| 23:01.10 | kresp0 | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 23:01.10 | kresp0 | chi*CLI> core show help sip show |
| 23:01.10 | kresp0 | No such command 'sip show'. |
| 23:01.23 | kresp0 | ??? |
| 23:01.29 | sawgood | sip show peers |
| 23:01.46 | kresp0 | No such command 'sip show peers' |
| 23:02.00 | p3nguin | You are have no chan_sip.so loaded. |
| 23:02.07 | kresp0 | Asterisk 1.6.2.9-2+squeeze3 |
| 23:02.13 | kresp0 | thank you p3nguin |
| 23:03.18 | _-Jon-_ | Oh sure, flame me, but not him! |
| 23:03.19 | _-Jon-_ | lol |
| 23:07.22 | kresp0 | p3nguin, im trying to load the module but no luck: |
| 23:07.23 | kresp0 | chi*CLI> load chan_sip.so |
| 23:07.23 | kresp0 | No such command 'load chan_sip.so' (type 'core show help load chan_sip.so' for other possible commands) |
| 23:07.51 | p3nguin | You're doing it wrong. |
| 23:07.56 | p3nguin | module load <module name> |
| 23:08.00 | kresp0 | yes, sure :D |
| 23:08.15 | kresp0 | thank you again! |
| 23:09.18 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 23:09.22 | cj | moo |
| 23:09.40 | cj | bunnies hopping around the floor |
| 23:09.42 | p3nguin | seri: After inspecting the nat translations port numbers, I think I have my answer. |
| 23:09.44 | cj | I have pics. |
| 23:10.12 | p3nguin | seri: I don't detect any ports that would be SIP on a non-standard port. |
| 23:10.25 | cj | have any of you used google voice for outgoing calls? |
| 23:10.41 | p3nguin | Yes. |
| 23:11.44 | cj | I plan on registering with my asterisk box from my android using Bria™ and then placing calls via the google voice account I've already set up on the machine |
| 23:12.02 | cj | p3nguin: got an example config I can look through? |
| 23:12.28 | p3nguin | It's in the wiki. |
| 23:12.32 | cj | I expect I need to add a context to my dial plan, eh? |
| 23:12.42 | p3nguin | You don't necessarily have to. |
| 23:12.54 | *** join/#asterisk bdfoster_ (~bdfoster@unaffiliated/bdfoster) |
| 23:12.55 | p3nguin | I've seen systems with everything crammed into a single context. |
| 23:13.05 | cj | oh, I plan on calling the PSTN via the google voice account, not just other google voice accounts |
| 23:13.36 | p3nguin | Of course. |
| 23:14.07 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:14.54 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 23:23.46 | cj | okay, that was almost right but instead wrong. |
| 23:24.31 | cj | I called my home line via google voice. when I picked up, everything was echoed back to on my home phone and no audio came back to my sip phone |
| 23:24.43 | cj | s/to on/to me on/ |
| 23:27.16 | *** join/#asterisk Eitan (~Eitan@12.192.84.98) |
| 23:27.41 | Eitan | anybody have any expereince porting In any numbers to ATT... wondering if i am going to have downtime even if my system is all set up.... |
| 23:28.10 | *** join/#asterisk faktorqm (~faktorqm_@190.244.153.123) |
| 23:28.25 | faktorqm | Hello! |
| 23:28.38 | p3nguin | There's usually no down time if both new and old systems are ready to go. |
| 23:29.15 | leifmadsen | no, porting is pretty much instantly available in my experience once the port actually happens (takes a few days) |
| 23:29.52 | faktorqm | I want to install a helpdesk ticketing system totally integrated with asterisk, but I don't have idea which software has this features |
| 23:29.54 | p3nguin | The last port I did (away from AT&T) happened and I didn't even know it was done. |
| 23:30.07 | leifmadsen | "totally integrated" is vague |
| 23:30.12 | p3nguin | I was prepared, so when it happened, it was seamless. |
| 23:30.44 | faktorqm | Do you know one? Here http://www.elastix.org/component/kunena/25/43856/ I found OTRS http://otrs.org/products/otrs-platform |
| 23:31.03 | leifmadsen | ~questions |
| 23:31.03 | infobot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
| 23:31.07 | faktorqm | but in the OTRS doesn't mention any asterisk related feature |
| 23:31.09 | leifmadsen | hmmm not what I wanted |
| 23:31.17 | p3nguin | haha |
| 23:31.19 | Eitan | p3nguin... thanks: ill make sure ATT porivsioning teams knows whats happening and make sure new DID's are set up on my system |
| 23:31.25 | Eitan | so when it switches over its working fine |
| 23:31.47 | leifmadsen | I still don't know what "totally integrated" means |
| 23:31.59 | leifmadsen | I'm just going to go with, "no, it doesn't exist" |
| 23:32.11 | p3nguin | It probably doesn't. |
| 23:32.43 | p3nguin | Maybe if the exact functionality was described... |
| 23:32.49 | leifmadsen | ya, that |
| 23:33.17 | faktorqm | leifmadsen: means, for example, if extension 102 calls to 103, 102 is the user, and 103 is the helpdesk guy, in the screen of that guy, show up a window with the last 5 tickets of that person |
| 23:33.17 | p3nguin | Asterisk can certainly receive and make phone calls. It can even make phone calls without having a phone to initiate the calls. |
| 23:33.18 | leifmadsen | I'm sure it could be built |
| 23:33.47 | leifmadsen | faktorqm: sure, that has nothing to do with the ticketing system though |
| 23:34.13 | p3nguin | Asterisk can do sql lookups. |
| 23:34.26 | hardwire | funcy ones. |
| 23:34.27 | p3nguin | And I'd assume you store ticket numbers in a database. |
| 23:34.40 | p3nguin | funcy? funky/fancy? |
| 23:35.01 | faktorqm | yes, in the database of a helpdesk program |
| 23:35.13 | p3nguin | That part will not be relevant. |
| 23:35.22 | hardwire | p3nguin: func_odbc funcy. |
| 23:35.41 | faktorqm | my question is, what helpdesk program has the feature to work with asterisk to do this kind of things? |
| 23:35.47 | p3nguin | Your datbase connector will not care HOW the ticket numbers were written, only that they exist. |
| 23:36.11 | p3nguin | So pick a ticket system that uses an open databse. |
| 23:36.29 | p3nguin | It won't "work with asterisk." |
| 23:36.30 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 23:36.45 | p3nguin | It will just work, and asterisk will look at the database it uses. |
| 23:36.52 | p3nguin | There's no integration of the two. |
| 23:38.18 | leifmadsen | ya, it really has nothing to do with the ticketing system at all |
| 23:38.22 | leifmadsen | just how the data is stored |
| 23:38.41 | p3nguin | I realize this isn't real sql, but: SELECT * where `uid` = $calleridnumber |
| 23:39.02 | p3nguin | That would be the basic idea. |
| 23:39.05 | faktorqm | ohhh I realized that! the helpdesk software is independent. I only need to see how the ticket data is stored by the helpdesk program |
| 23:39.12 | leifmadsen | right |
| 23:39.17 | faktorqm | excellent |
| 23:39.17 | leifmadsen | and you want to access it with func_odbc |
| 23:39.26 | faktorqm | thank you very much for your help |
| 23:39.54 | faktorqm | regards!! |
| 23:39.56 | *** part/#asterisk faktorqm (~faktorqm_@190.244.153.123) |
| 23:40.11 | [TK]D-Fender | HOW I CAN EVERYTHING?!?! |
| 23:40.19 | p3nguin | >: |
| 23:41.03 | sawgood | teach me too please! |
| 23:41.30 | [TK]D-Fender | Funnier with 1 less "o" :) |
| 23:41.30 | hardwire | I was about to suggest using curl |
| 23:41.51 | hardwire | now I just feel like he used p3nguin |
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