| 00:16.57 | *** join/#asterisk alx0 (~alx@213.138.88.177) |
| 00:20.02 | *** join/#asterisk De_Mon (47b42a1d@gateway/web/freenode/ip.71.180.42.29) |
| 00:21.16 | De_Mon | Is it possible for a PRI/line card on an internet facing asterisk server to be hijacked by a hacker to make calls without going to through the pbx software? |
| 00:21.28 | De_Mon | over the internet |
| 00:24.05 | carrar | sure |
| 00:25.09 | carrar | though I would assume they would use whatever software you have installed aka asterisk |
| 00:25.37 | carrar | otherwise asterisk probably would not be able to use the card as well |
| 00:26.25 | carrar | setup one of your PRI DID's to allow someone to turn around and dial out |
| 00:26.51 | De_Mon | you would use some driver to give the system access to the card |
| 00:27.18 | De_Mon | and you would need software connecting the incoming traffic from the internet to the local driver |
| 00:27.24 | carrar | You could shim some driver between asterisk's driver and the card but that seems overly compleex for the return |
| 00:27.50 | carrar | So I would say probably not |
| 00:28.00 | De_Mon | But something has to be listening for the incoming connection from the internet and then giving it access to the card |
| 00:28.12 | carrar | sure |
| 00:28.18 | carrar | lsof is your friend |
| 00:28.33 | De_Mon | and if the connection is coming from port 5060, and asterisk is the only thing listening on port 5060, wouldn't asterisk have to allow the interaction? |
| 00:28.42 | De_Mon | sorry, *going to* 5060 |
| 00:28.59 | carrar | then it would be asterisk |
| 00:29.25 | carrar | more then likely you have a miss configured server and poeple are exploiting it |
| 00:29.34 | carrar | aka misconfigured asterisk |
| 00:30.14 | De_Mon | yeah... we have a switchvox appliance (don't ask) and digium support is saying this type of behavior is normal and there is nothing we can do except block the attackers ip address |
| 00:30.28 | De_Mon | My bullshit detector is going crazy and just wanted to confirm |
| 00:30.36 | carrar | heh |
| 00:30.37 | carrar | yeah |
| 00:30.53 | carrar | with switchvox block all and only allow the known networks |
| 00:31.04 | carrar | which sucks for roaming |
| 00:31.32 | carrar | Or put something infront of it that can detect abuse |
| 00:32.47 | De_Mon | wait, are you saying this is not a configuration issue with the switchvox? |
| 00:33.15 | carrar | what are we talking about? |
| 00:33.22 | carrar | You started out talking about asterisk |
| 00:33.26 | carrar | now you are talking about switchvox |
| 00:33.41 | carrar | pick one |
| 00:33.53 | De_Mon | sorry... I thought switchvox appliance uses asterisk under the hood |
| 00:34.04 | carrar | it does but you do not get to touch it |
| 00:35.05 | carrar | but likewise, you can miss configure switched also |
| 00:35.20 | carrar | if you don't understand the dialplan you are creating |
| 00:35.36 | carrar | s/switched/switchvox/ |
| 00:35.50 | De_Mon | We are being told the switchvox appliance is not involved with this call being made because it's going straight from the internet to the line card |
| 00:36.12 | De_Mon | this this a thing, or is he blowing smoke |
| 00:37.00 | carrar | what is the "line card" |
| 00:37.52 | carrar | is it a physical card inside the SwitchVox server AKA a digium T1 card? |
| 00:40.23 | De_Mon | yeah |
| 00:40.48 | De_Mon | a TE133, it's a physical card |
| 00:41.04 | carrar | then that is part of the switchvox applianc |
| 00:41.35 | carrar | whoever configured your switchvox server is responsible for this |
| 00:41.55 | carrar | would be my guess |
| 00:42.34 | De_Mon | thats what I'm thinking, and I can't figure out why on earth digium support would say something so outlandish |
| 00:42.47 | De_Mon | it was so crazy I thought it might just be true... |
| 00:42.56 | carrar | Sounds like a communication missunderstanding |
| 00:49.00 | phix | would an asus eeebox (1.2Ghz cellery, dual core, 4Gb RAM) be a good Asterisk server for 12 SIP phones, 4 concurrent lines? |
| 00:49.27 | [TK]D-Fender | My watch can handle that ... and it's ANALOG |
| 00:49.39 | [TK]D-Fender | heads out for the night... |
| 00:49.48 | phix | [TK]D-Fender: You were answering my question? |
| 00:49.55 | [TK]D-Fender | phix: yes |
| 00:50.04 | phix | ah, awesome ;) |
| 00:50.07 | phix | ok have a good night! :) |
| 00:50.21 | phix | Don't drink and drive! |
| 01:19.10 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 01:23.56 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:27.41 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:28.42 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:30.06 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:30.49 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:32.16 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:32.17 | De_Mon | thanks for the advise carrar |
| 01:33.14 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:34.01 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:46.34 | *** join/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 01:48.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:51.38 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 01:53.20 | volga629 | Hello Everyone, what possible cause that client with TLS losing registration each 2 min |
| 02:04.34 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 02:13.54 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 02:15.57 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 02:28.27 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 02:50.52 | *** join/#asterisk tensai (~quassel@174-126-39-53.cpe.cableone.net) |
| 03:11.55 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 03:36.16 | *** part/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 04:01.36 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 04:08.14 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 04:12.54 | *** join/#asterisk pfui (c52312f0@gateway/web/freenode/ip.197.35.18.240) |
| 04:17.05 | pfui | trying to send a sip message via asterisk but I keep getting errors on the console that "function MESSAGE not registered" and "No application MessageSend for extension", what could be the problem? |
| 04:27.41 | Penguin | No application MessageSend for extension <----- This is the problem. |
| 04:27.57 | pfui | hmm? |
| 04:35.53 | pfui | okay, got it. |
| 04:35.56 | pfui | thanks. |
| 04:44.08 | *** join/#asterisk russum (~russum@ool-4a599f87.dyn.optonline.net) |
| 04:44.46 | *** part/#asterisk russum (~russum@ool-4a599f87.dyn.optonline.net) |
| 04:47.40 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 05:24.40 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 05:41.08 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 06:09.17 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:11.05 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 06:42.51 | *** join/#asterisk CeBe (~CeBe@port-92-206-220-240.dynamic.qsc.de) |
| 06:46.04 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.244.113) |
| 06:47.01 | *** join/#asterisk jploh (~textual@49.144.25.156) |
| 06:58.38 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 07:00.29 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:13.32 | *** join/#asterisk jsjc (~Adium@0.Red-2-136-89.dynamicIP.rima-tde.net) |
| 07:21.55 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:14.45 | saxa | hi , what hardware should i buy to be able to get R2D signalling working |
| 08:15.04 | saxa | they told me I need a card with R2D signalling support |
| 08:17.50 | WIMPy | What is that? |
| 08:18.35 | saxa | hi WIMPy is the Brasilian newer version os ISDN |
| 08:18.46 | saxa | i mean, wht they use in Brasil |
| 08:19.01 | saxa | they said its R2D |
| 08:19.16 | saxa | but I think its MFC/R2 |
| 08:19.20 | saxa | or something similar |
| 08:19.22 | WIMPy | New? They no longer use bog standard DSS1? |
| 08:19.32 | saxa | have no idea |
| 08:19.47 | WIMPy | MFC is pre ISDN. |
| 08:20.03 | WIMPy | What kind of line is this? |
| 08:21.43 | WIMPy | You should probably make sure you get an ISDN line. Then yu can use any hard- and software you like. |
| 08:22.01 | saxa | ok, they said to me that is R2D but I have no idea, they call it also Digitronco |
| 08:22.34 | saxa | so now I have a TDM410P analog card, but I need to buy a new one to get this kind of line working |
| 08:22.57 | saxa | and I have no idea what they use , it is something standard in Brasil AFAIK |
| 08:23.45 | WIMPy | Standard lines should be available there. Should make things easier. |
| 08:25.25 | saxa | AFAI looked around, this are probably E1 cards |
| 08:25.37 | saxa | but I'm not sure if this is it. |
| 08:26.01 | WIMPy | Yes. Looks like E1, but NFI what R2D might be. |
| 08:26.32 | WIMPy | But I also saw "2B+D+R2D". |
| 08:27.22 | saxa | ok this is something what I will probably get |
| 08:27.25 | saxa | http://www.elastixbrasil.com.br/index.php?option=com_fireboard&Itemid=74&func=view&catid=5&id=2799 |
| 08:27.42 | saxa | the configuration needed will probably be this |
| 08:28.39 | WIMPy | So it's a variation of C5? |
| 08:29.11 | WIMPy | I'd try to change that to an ISDN line if possible. |
| 08:30.45 | saxa | seems people uses digivoice VB3030 e VB0408O |
| 08:33.34 | saxa | WIMPy: i asked if they have an ISDN line but seems they do not have it |
| 08:33.50 | saxa | any suggestions on those digivoice cards ? |
| 08:40.18 | *** join/#asterisk jsjc (~Adium@227.Red-83-32-91.dynamicIP.rima-tde.net) |
| 08:45.37 | WIMPy | Not seen before. |
| 09:06.44 | saxa | ok thx |
| 09:08.27 | saxa | ok, basically which cards you suggest for a E1 line ? |
| 09:09.51 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 09:24.28 | *** join/#asterisk Albrecht (~Albrecht@e197163.upc-e.chello.nl) |
| 09:24.32 | Albrecht | hi all |
| 09:30.48 | saxa | WIMPy: ok, Digitronco is E1 it has 32 channels of 64kbps |
| 09:31.43 | saxa | it uses 30 channels for voice, one for signalling and one for sync |
| 09:32.20 | saxa | Digitronco is a commercial name in Brasil from the operator Oi |
| 09:32.39 | saxa | but it is know as E1 link |
| 09:34.58 | saxa | http://www.telemarnetwork.com.br/CfxTemp/GUI/ArquivosUpload/167versaoImpr.pdf unfortunately this is only in Portugues |
| 09:39.37 | saxa | so thats it, now, knowing it is a E1 i will get, are there any suggestions for a card I have to buy ? |
| 09:44.41 | *** join/#asterisk bipul (31c95c72@gateway/web/freenode/ip.49.201.92.114) |
| 09:45.36 | *** part/#asterisk bipul (31c95c72@gateway/web/freenode/ip.49.201.92.114) |
| 09:47.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.204) |
| 10:06.59 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 10:12.18 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginmedia.com) |
| 10:18.09 | *** join/#asterisk kresp0 (~kresp0@136.Red-95-120-50.dynamicIP.rima-tde.net) |
| 10:18.26 | *** join/#asterisk user258467 (~user25846@cue83-1-88-163-202-186.fbx.proxad.net) |
| 10:19.26 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
| 10:19.36 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.204) |
| 10:21.00 | Ice_Strike | Morning |
| 10:22.59 | Ice_Strike | What is your ip scheme for workstations and phones? |
| 10:24.20 | user258467 | Hi , i have a problem with my server it plays the closed audio message how could I debug it |
| 10:39.46 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 11:01.02 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 11:22.34 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 11:23.04 | saxa | it seems that for my case an 1TE122B card should be ok |
| 11:23.35 | saxa | i think the best solution for using it with asterisk would be Digium correct ? |
| 11:23.43 | *** join/#asterisk barbosa2 (~juliano.b@177-069-248-123.static.ctbctelecom.com.br) |
| 11:23.47 | saxa | or is it better to go with some other brand ? |
| 11:23.54 | saxa | Alternatives ? |
| 11:27.26 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 12:00.29 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 12:19.39 | *** join/#asterisk gigiozzo (02ee5811@gateway/web/freenode/ip.2.238.88.17) |
| 12:19.43 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 12:22.45 | gigiozzo | i guys, i have an issue with my asterisk config, i forward incoming call from one asterisk server to another via sip, all seems working but after around 300 seconds (5 minutes) call is closed |
| 12:25.02 | gigiozzo | anybody can help me? |
| 12:37.31 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 12:40.33 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 12:54.18 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 13:30.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 13:49.28 | saxa | hi any suggestion for a cheap single line E1 card ? |
| 13:49.58 | saxa | i was thinking about a TE122, but I see there is already a TE133 |
| 13:50.21 | saxa | but are there any alternatives well supported in linux/asterisk ? |
| 13:52.14 | saxa | i saw there is plenty of options in China, with various producers |
| 13:52.32 | saxa | but are those supported by libpri and dahdi ? |
| 13:55.34 | saxa | i see openr2 protocol is sponsored by sangoma, maybe is better to choose a sangoma card over an digium ? |
| 14:52.14 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 14:52.27 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:08.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:09.19 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:09.29 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 15:26.34 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:27.15 | *** join/#asterisk evilman_home (kvirc@89.179.77.66) |
| 15:28.49 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 15:29.09 | qakhan | hi all. is correct? |
| 15:29.10 | qakhan | Set(day=$["${day}"="01" | "${day}"<="31"]) |
| 15:32.39 | [TK]D-Fender | qakhan: Does it work? |
| 15:32.54 | qakhan | no |
| 15:33.27 | [TK]D-Fender | Show us |
| 15:34.01 | qakhan | i want to set a veriable with values as = 01 and <= 31 |
| 15:34.11 | qakhan | Set("SIP/5288-00000035", "day=1") |
| 15:34.40 | [TK]D-Fender | qakhan: You aren't showing us what the value holds at the start |
| 15:35.05 | [TK]D-Fender | qakhan: Do you still not understand what COMPLETE backup means? |
| 15:35.16 | [TK]D-Fender | qakhan: NoOP() the variable FIRST so we can see what it holds |
| 15:35.24 | [TK]D-Fender | qakhan: and PASTEBIN the call |
| 15:37.35 | qakhan | here http://pastebin.com/beQCzVnB |
| 15:38.19 | [TK]D-Fender | qakhan: I just told you to NoOp it BEFORE your test |
| 15:40.23 | qakhan | [TK]D-Fender check this one |
| 15:40.24 | qakhan | http://pastebin.com/NU5xPkKV |
| 15:40.59 | [TK]D-Fender | qakhan: You're still not listening |
| 15:41.06 | [TK]D-Fender | NoOp the DAY before your expression |
| 15:41.43 | [TK]D-Fender | qakhan: You asked if that expression that uses it it good. And you still aren't displaying that specific variable BEFORE the test like I've told you repeatedly now. |
| 15:42.02 | [TK]D-Fender | qakhan: Don't ask if the formula makes sense if you aren't looking at what you are giving it |
| 15:45.18 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 15:46.02 | qakhan | [TK]D-Fender we can see in cli what i was giving in day. anyhow please see this http://pastebin.com/vCQj8mkX |
| 15:48.02 | [TK]D-Fender | qakhan: So -- Executing [5101@outgoing:4] Set("SIP/5288-00000037", "day=26") in new stack |
| 15:48.07 | [TK]D-Fender | qakhan: day = 26 |
| 15:48.19 | [TK]D-Fender | qakhan: exten => 5101,n(jandate),Set(day=$["${day}"="01" | "${day}"<="31"]) <-- your expression |
| 15:48.35 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 15:48.37 | [TK]D-Fender | qakhan: -- Executing [5101@outgoing:7] Set("SIP/5288-00000037", "day=1") in new stack <-- the result |
| 15:48.51 | qakhan | yes |
| 15:48.51 | [TK]D-Fender | qakhan: Test does expectly what it says. |
| 15:49.09 | [TK]D-Fender | qakhan: 26 IS less than or equal to 31. |
| 15:49.22 | [TK]D-Fender | qakhan: So it returns "1" |
| 15:49.45 | [TK]D-Fender | qakhan: -- Executing [5101@outgoing:7] Set("SIP/5288-00000037", "day=1") in new stack <--which is what it gets set to. |
| 15:53.44 | qakhan | Thank you [TK]D-Fender you are always helpful :) |
| 15:54.37 | [TK]D-Fender | qakhan: So day started with 26 ... and then you replace that with the answer to "is it smaller than 32?" |
| 15:55.02 | [TK]D-Fender | qakhan: Now I don't know WHY you would put that answer into a variable called "day" |
| 15:56.13 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:56.41 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginmedia.com) |
| 16:00.40 | qakhan | i changed the veriable name to chkday from day |
| 16:02.24 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 16:03.26 | [TK]D-Fender | qakhan: And what are you doing with that? |
| 16:09.44 | *** join/#asterisk mKn0wt (~mtv@190.181.143.158) |
| 16:33.07 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 16:39.53 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:54.42 | *** join/#asterisk umpa_ (~ubuntu@ec2-54-200-75-10.us-west-2.compute.amazonaws.com) |
| 17:57.24 | *** part/#asterisk umpa_ (~ubuntu@ec2-54-200-75-10.us-west-2.compute.amazonaws.com) |
| 17:58.11 | *** join/#asterisk umpa_ (~ubuntu@ec2-54-200-75-10.us-west-2.compute.amazonaws.com) |
| 17:59.26 | umpa_ | k |
| 18:01.18 | *** part/#asterisk umpa_ (~ubuntu@ec2-54-200-75-10.us-west-2.compute.amazonaws.com) |
| 18:04.46 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 18:20.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:26.41 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 18:34.17 | phix | ump is the worst gun in CSS, I am suprised it is in CS: GO |
| 18:37.04 | phix | sure it is a .45 ACP but seriously, I would prefer a MP5 regardless if it is more expensive |
| 18:37.27 | phix | (which the Mp5 is) |
| 18:39.42 | phix | so in conclusion, umpa_ should be now known as mp5a_ |
| 18:39.56 | phix | or p90a_ |
| 18:41.50 | ChannelZ | I'll take them all |
| 18:44.34 | phix | ChannelZ: mp5 is meh, bison is yeah! and p90 is w00000!!!!! |
| 18:44.43 | phix | ump is fail |
| 18:47.39 | phix | Belgiums > Russians |
| 18:48.12 | phix | <PROTECTED> |
| 18:48.32 | phix | s |
| 19:41.30 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:45.23 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:46.58 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:49.05 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 20:00.56 | *** join/#asterisk RudyValencia (~self@unaffiliated/rudyvalencia) |
| 20:01.13 | RudyValencia | Hi I'm trying to set up asterisk 11.5.1 and I can't seem to get clients to connect with tls |
| 20:01.42 | RudyValencia | any suggestion as to how to get tls working properly for any client (some of which are road warrior/roaming clients)? |
| 20:16.17 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 20:22.57 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 20:30.44 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 20:32.36 | ChannelZ | Ugh. UPS is worthless. |
| 20:36.07 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 21:07.49 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:09.21 | danfromuk | Hi. If I need to store channel related data in the asterisk database, will it cause a problem if i dont delete it? |
| 21:09.24 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 21:10.40 | danfromuk | the data thats added only needs to be kept during the active call |
| 21:20.13 | Penguin | Maybe. It will remain there until it is deleted or overwritten. |
| 21:20.48 | danfromuk | does a restart delete asterisk db data? or does it remain? |
| 21:23.57 | Penguin | It remains. That's the point of the db -- it is persistent. |
| 21:24.09 | ChannelZ | like bad breath |
| 21:25.14 | danfromuk | Has anyone done any testing on the maximum number of records that can be stored in the asterisk db? |
| 21:25.28 | danfromuk | Or any advice on the limitations? |
| 21:25.56 | danfromuk | Just trying to get around the channel variables getting lost during an attended transfer. |
| 21:26.23 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 21:26.28 | Penguin | You're doing it wrong. |
| 21:27.06 | danfromuk | After a very long discussion about this issue a month ago, it was concluded that i have to use a key and store the data in astdb |
| 21:27.10 | Penguin | The astdb will hold lots of records. I doubt you have so little RAM that you can fill up the db. |
| 21:27.53 | Penguin | I doubt that is accurate. You just aren't setting your variables correctly to have them be inherited to a newly created channel. |
| 21:28.51 | danfromuk | Penguin: Trust me. I displayed my dialplans and the conclusion was that when a channel is transferred through an attended transfer, the original channel variables are unavailable. Even if __ is used |
| 21:29.02 | Penguin | That's bizarre. |
| 21:29.13 | Penguin | But if it is true, the astdb is the way to make it work. |
| 21:29.16 | danfromuk | As I recall, [TK]D-Fender was very vocal on the subject |
| 21:29.35 | Penguin | Usually setting with underscores solves inheritance problems. |
| 21:30.22 | Penguin | And the variable will always be overwritten every time the channel sets it. Have a stale variable shouldn't be a problem if the channel is always setting it. |
| 21:34.46 | ChannelZ | if you can't get channel variables to pass through the sequence of your call, how are you going to track the rows in the database any better? |
| 21:34.54 | [TK]D-Fender | Inheritance happens when you DIAL another channel |
| 21:35.24 | Penguin | To do an attended transfer, you must DIAL another extension. |
| 21:35.27 | [TK]D-Fender | When you do a blind transfer, you take that channel, with all it's vars and continue execution of dialplan somewhere else. |
| 21:35.31 | Penguin | its |
| 21:35.51 | [TK]D-Fender | An attended transfer starts as a completely separate channel and ony gets audio bridged later. Variables are not magically carried over |
| 21:37.19 | danfromuk | The original channel name is the key. The other channels, even if the call is transferred through an attended transfer, have the BRIDGEPEER variable which matches the original channel name |
| 21:37.34 | danfromuk | Not tried a blind transfer yet though |
| 21:40.40 | Penguin | If we assume there is no way normal channel variables can carry over, use of the astdb is a great way to go around the problem. |
| 21:41.34 | danfromuk | can the key contain a '/' ? |
| 21:41.43 | [TK]D-Fender | danfromuk: yes |
| 21:42.48 | danfromuk | thanks |
| 21:44.45 | Penguin | Can I run Asterisk on Plan 9? |
| 21:55.28 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 22:04.16 | *** join/#asterisk xzarth_ (~krikkit@dh207-28-49.xnet.hr) |
| 22:17.32 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:00.19 | *** join/#asterisk russum (~russum@ool-4a599f87.dyn.optonline.net) |