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| 06:37.53 | stev3n | hi, is it possible to make asterisk work with sip-trunk as follows: on the 1st line of talk, and hung up, the next call should be to the 2nd line, and so on around the circle? |
| 06:40.33 | stev3n | or line to the SIP trunks are not numbered? |
| 06:40.49 | stev3n | *on the SIP trunks |
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| 06:44.48 | [TK]D-Fender | You can set this up if you feel like it |
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| 06:45.15 | [TK]D-Fender | That's what the dialplan is for.. its your rules... set up a persistent value to track the next one to use and process away |
| 06:47.08 | stev3n | how to number lines in the sip-trunk for use in dialplan? |
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| 06:50.08 | [TK]D-Fender | However you want. just keep a counter for which "logical" number you'r on. |
| 06:51.44 | [TK]D-Fender | If you have 3 SIP peers it doesn't matter if you named them [tom], [dick], and [harry]. Keep a global variable, or AstDB value for whic "number you're on. When they go to dial out, check the number and bump it up one, or rotate back to 1. use the peer based on the number you're at |
| 06:59.45 | stev3n | yes, but I dont't find example, how to adress specific line in sip-trunk |
| 07:01.06 | kaldemar | what do you mean by sip-trunk? |
| 07:02.18 | [TK]D-Fender | What is a LINE ins a "sip-trunk"? You are using vague and inappropriate terminology. |
| 07:02.20 | stev3n | I mean physical line from provider |
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| 07:03.09 | kaldemar | stev3n: what technology are you using to dial out? |
| 07:03.27 | [TK]D-Fender | As for samples... this s dialplan we're talking about. It's PROGRAMMING. You aren't going to find samples for every little idea. You've got the basic building blcoks with channel variables (including a GLOBAL you'll need in this case to track where you are. The rest if Gotoif's, set's, etc |
| 07:04.38 | stev3n | now i'm using sip, and asterisk use free sip channels to dial out |
| 07:06.13 | [TK]D-Fender | counter = counter + 1 |
| 07:06.42 | [TK]D-Fender | if counter =6 then set counter = 1 |
| 07:07.09 | [TK]D-Fender | if counter = 1 then dial(sip/john/1234567890) |
| 07:07.17 | [TK]D-Fender | if counter = 1 then dial(sip/fred/1234567890) |
| 07:07.23 | kaldemar | unless you have multiple peers in sip.conf you want to bundle together, you don't need to do anything. just dial. |
| 07:07.26 | [TK]D-Fender | if counter = 2 then dial(sip/fred/1234567890) (correction) |
| 07:07.39 | [TK]D-Fender | if counter = 3 then dial(sip/mark/1234567890) (correction) |
| 07:07.53 | [TK]D-Fender | Warning, not ACTUAL dialplan.... but this is the basic concept |
| 07:08.47 | stev3n | [TK]D-Fender: thanks for the explain |
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| 07:12.04 | [TK]D-Fender | bed time, checking out, back tomorrow |
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| 08:53.41 | BQffen | I have a problem with my asterisk.conf, changing astctlpermissions does not seem to have any effect. Has anybody experienced this? |
| 08:56.18 | wdoekes | BQffen: are you trying to give more permissions or less? |
| 08:56.30 | wdoekes | you could be looking at umask limitations |
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| 08:57.46 | wdoekes | $ touch foo; umask 0007; touch bar; ls -l |
| 08:57.50 | wdoekes | -rw-rw---- 1 walter walter 0 Jan 29 09:57 bar |
| 08:57.50 | wdoekes | -rw-rw-r-- 1 walter walter 0 Jan 29 09:56 foo |
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| 09:06.16 | BQffen | wdoekes: I need another user to be able to convert a recorded file using sox, for some reason this requires +x |
| 09:06.50 | BQffen | but changing 0600 to 0750 does not change anything |
| 09:11.03 | BQffen | default is 0660 i guess, i did som testing |
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| 09:15.51 | wdoekes | astctlpermissions has nothing to do with the writing of audio files |
| 09:18.28 | wdoekes | #define AST_FILE_MODE 0666 |
| 09:18.52 | wdoekes | so I guess you have an umask that drops the o perms |
| 09:19.16 | wdoekes | and no, another user does not need +x |
| 09:19.45 | wdoekes | another user needs o+r on the file (and o+x on the directory) |
| 09:20.20 | wdoekes | BQffen: you could simply System(chmod 777 your_recorded_file.wav) in your dialplan |
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| 11:20.51 | mirela666 | Hi, is there any way to rewrite the calleid to original caller when forward request is sent? |
| 11:21.14 | mirela666 | for example: A calling B and B transfering to C |
| 11:21.37 | mirela666 | on the phone of C I want callerID of A to be displayed, and not B |
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| 11:22.39 | WIMPy | That would require B to use a blind transfer. |
| 11:23.19 | WIMPy | On an attended Transfer it should update after the transfer is completed. |
| 11:23.49 | mirela666 | yes i think it is a blind |
| 11:23.56 | mirela666 | yes defenatly |
| 11:24.28 | mirela666 | so the blind trasfer can't be updated? |
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| 11:56.13 | mccar | Hi! Can anybody help with AsteriskNOW 3.0? I have problem with NAT configuring |
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| 11:59.47 | mccar | Asterisk version is 11.2.1. I've got a client behind NAT. When I call from one client(NAT) to another(not NAT) voice appear only after 10-30 seconds. I can see in WireShark that Asterisk send RTP to my external IP and then to client's NAT device |
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| 12:05.25 | sekil | asterisk 11.x is preferred stable version these days |
| 12:05.26 | sekil | ? |
| 12:05.43 | WIMPy | yes |
| 12:05.50 | sekil | k |
| 12:06.23 | eirirs | 1 |
| 12:06.50 | mirela666 | sekil: LTS releases from the official download site are considered as stable I guess |
| 12:07.17 | mccar | No one able/want to help&// |
| 12:07.21 | mccar | ? |
| 12:07.23 | sekil | hm..there is two of them ..or more.. |
| 12:07.32 | sekil | so I wanted to ask about what to use if in doubt |
| 12:09.22 | mirela666 | I suggest: 11.6-cert1 or 1.8.15-cert4 |
| 12:09.41 | mirela666 | http://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
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| 13:01.19 | madduck | Do people have experience with solutions that allow for flexible, dynamic association management between incoming call sources, telephones and users? I.e. I want all our telephones (in various offices) to be registered at all times, but I want a flexible, scriptable way to associate incoming numbers with those devices depending on where users are. |
| 13:06.25 | mccar | Кто-нибудь понял что ему надо? |
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| 13:54.45 | chees0 | quick question: Is there any way to make apt-get not flip out when installing asterisk 12? |
| 13:57.03 | chees0 | i'm running debian jessie |
| 13:58.21 | skrusty | use src? :) |
| 13:58.37 | chees0 | apt-get will try to downgrade after that |
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| 13:58.46 | chees0 | but good thinking :) |
| 13:58.48 | skrusty | tbh, i use debian and much prefer to compile from src |
| 13:58.57 | skrusty | (for asterisk) |
| 14:00.20 | chees0 | the problem is not asterisk by itself, but the libraries. there's tons of library versions that have to be frozen after compiling asterisk manually |
| 14:01.21 | skrusty | that may be so, but i've never done that myself and never had an issue yet (that i know of :) |
| 14:01.30 | skrusty | but point taken |
| 14:05.56 | chees0 | i've already given up for now, so no worries. trying the 11 route next |
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| 14:23.32 | Katty | morning fender bender. |
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| 14:49.20 | n3hxs | Morning Katty how are you ? |
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| 15:18.06 | mirela666 | hey, is there any way to rewrite callerid on blind transfer to original callerid |
| 15:18.24 | mirela666 | for example A clling B and B transfering to C |
| 15:18.25 | [TK]D-Fender | a blind transfer IS the original callerid |
| 15:18.37 | mirela666 | so C can see A in from header |
| 15:18.51 | [TK]D-Fender | A calls B. B blind-transfers to C. C sees "A" |
| 15:19.06 | mirela666 | aha then it is for attended |
| 15:19.14 | mirela666 | i mixed up |
| 15:19.43 | mirela666 | ok I will have to preform test and see it myself |
| 15:19.48 | mirela666 | thx TK |
| 15:20.00 | [TK]D-Fender | Attended is deliberately "B". That's so C doesn't reflexively go "I don't want to talkto anyone" and instead goes "Oh... it's my assistant... they know I'm busy so it must be important..." |
| 15:22.29 | Katty | n3hxs: just peachy so far! how're you dear? |
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| 15:52.53 | mccar | Hi! Can anybody help with SIP MESSSAGE behind NAT? |
| 15:55.06 | pabelanger | ~ask |
| 15:55.06 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:55.11 | pabelanger | mccar, ^ |
| 15:57.07 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 16:00.08 | mccar | Aserisk(192.168.100.251) <(VPN)> local network(192.168.2.0/24) . In local network I have Wifi router which has its own local network (192.168.1.0/24) I try to send sip messages from Bria(Iphone) to X-lite on local machine. Local machine send messages to WiFi router IP but I can't see then on IPhone |
| 16:00.20 | mccar | Too complicated?.. |
| 16:01.11 | pabelanger | ~book |
| 16:01.12 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:01.29 | pabelanger | mccar, there is a section about sip and nat, all your questions should be answered |
| 16:01.38 | pabelanger | basically SIP and NAT are bad for each other. |
| 16:01.51 | pabelanger | You'll need to setup localnet in sip.conf and externip too |
| 16:02.10 | mccar | Yes, I know, but Iphone can't work without Wifi |
| 16:02.26 | mccar | I did it |
| 16:02.52 | pabelanger | So what is the issue? |
| 16:03.27 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:04.36 | [TK]D-Fender | mccar: SHOW us the problem |
| 16:04.38 | [TK]D-Fender | ~pb |
| 16:04.38 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:04.41 | [TK]D-Fender | ^^^ |
| 16:04.59 | mccar | Iphone registered pn Asterisk like 192.168.2.98:57473 It's nat. When I send messgae from X-lite to Iphone I see in Wireshark that messages are sent to 192.168.2.98:57473 but masseges are not appear |
| 16:05.30 | [TK]D-Fender | Asterisk SIP debug + verbose, not Wireshark |
| 16:05.43 | pabelanger | sounds like a routing problem, not asterisk |
| 16:05.47 | mccar | Just a minute |
| 16:09.26 | mccar | http://pastebin.com/uaUVJ7UW |
| 16:11.53 | [TK]D-Fender | mccar: it looks like you're specifying a peer... AND a URI |
| 16:12.28 | mccar | Can you explain, please |
| 16:12.39 | [TK]D-Fender | <PROTECTED> |
| 16:12.47 | [TK]D-Fender | look what you have right there in your dialplan |
| 16:15.03 | mccar | I took it "as is" from someones blog and unfortunately Im not understand what it mean... |
| 16:16.19 | [TK]D-Fender | I'd first read the documentation on these apps and other samples.... |
| 16:16.35 | [TK]D-Fender | "As-is" is not a good basis for expecttaion or debugging.... |
| 16:16.52 | mccar | You a right... |
| 16:17.18 | mccar | Here it is: http://pastebin.com/4pfyU9L9 |
| 16:20.21 | mccar | But there is no problems with that dialplan if I send message from one X-lite to another |
| 16:21.41 | [TK]D-Fender | at the same IP's? |
| 16:22.19 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 16:23.56 | mccar | No, from other networks connected by VPN |
| 16:24.23 | mccar | So I can call from IPhone |
| 16:24.32 | mccar | Issue only with messages |
| 16:25.50 | [TK]D-Fender | so a PC on that same network would look the same AND get the message and respond? |
| 16:28.57 | mccar | Yes, I can call and send messages from PC in my local net to another PC in anothe LAN. All networks has VPN with 192.168.100.0/24 |
| 16:30.25 | [TK]D-Fender | Then it sounds like a pure iphone problem... one consideration is that I've never overcome the WiFi SLEEP issue on them where an app can't prevent the iphone from shutting down power on wiFi |
| 16:32.18 | mccar | Maybe you are right, but I can make a call on iphone and it works fine... |
| 16:33.37 | mccar | And messages work if 2 iphones in same network |
| 16:50.42 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
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| 17:04.16 | ChannelZ-Wk | Hmm. Anyone know of some way to clear the 'Answered Calls' and 'Missed Calls' etc on Cisco/Linksys SPAs, besides deleting them one by one manually from the actual phone? A secret undocumented URL or anything? |
| 17:04.44 | [TK]D-Fender | ChannelZsee the power cord? *YANK* |
| 17:05.10 | ChannelZ-Wk | Unfortunately that doesn't work! |
| 17:05.41 | ChannelZ-Wk | I guess maybe if I did a 'factory reset' maybe it would |
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| 17:11.39 | Kobaz | where can i get a nice table listing of all premium rate number prefixes... so i can block them of course |
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| 17:19.45 | ChannelZ-Wk | Like 976? |
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| 17:31.37 | Kobaz | yeah i guess |
| 17:31.41 | Kobaz | anything that's not 900 |
| 17:31.45 | Kobaz | and still premium |
| 17:32.08 | Kobaz | isn;t there another one like 970 and 540 |
| 17:35.59 | ChannelZ-Wk | Well in the US I believe the whole of 9XX as an area code is.. not heard of 540 |
| 17:36.20 | ChannelZ-Wk | Actually that's not true. |
| 17:36.34 | ChannelZ-Wk | 970 is an area code in Colorado (my own state, hahah) |
| 17:37.23 | ChannelZ-Wk | http://en.wikipedia.org/wiki/Premium-rate_telephone_number |
| 17:39.20 | ChannelZ-Wk | Seems like 900 and 976. I so far haven't found anything definitive on NANP |
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| 17:43.57 | ruben231 | hi guys anyone here have working config somehow for fail2ban, mine i have setup does not block at all..any idea..? |
| 17:44.29 | Kobaz | ruben231: either you're not logging enough, or your log file is specified wrong in fail2ban, or your regexes in fail2ban are wrong |
| 17:44.30 | [TK]D-Fender | idea: show us what you're doing. Show us status dumps from fail2ban. show us log files |
| 17:44.41 | Kobaz | did i miss anything? |
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| 17:45.23 | Kobaz | ruben231: working config would depend on several things, version of fail2ban, and version of asterisk |
| 17:45.41 | Kobaz | ruben231: and also what you're tring to block in asterisk (sip/iax/etc) |
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| 17:47.45 | ruben231 | Kobaz: ok let me share my fail2ban config , im using asterisk 1.8 now |
| 17:48.04 | ChannelZ-Wk | 1.8 doesn't log all things you're likely to want IIRC |
| 17:48.05 | Kobaz | ruben231: and then also share asterisk logs containing failed attempt(s) you wish to block |
| 17:48.27 | Kobaz | ChannelZ-Wk: yeah, i had to add a bunch of extra logging to get fail2ban to behave the way i wanted |
| 17:49.02 | ChannelZ-Wk | If I remember registrations never showed the actual IP the traffic came from, I hacked that into my own to make that work.. |
| 17:49.11 | ChannelZ-Wk | in 10 it's part of the new security log though. |
| 17:49.23 | ChannelZ-Wk | And then if you're on 12 using pjsip the format changed slightly |
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| 18:01.02 | Kobaz | yeah |
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| 18:30.47 | *** mode/#asterisk [+o file] by ChanServ |
| 18:36.26 | ruben231 | Kobaz: hi there this is my fail2ban setting for asterisk 1.8 - http://pastebin.com/dPTpLNZ8 --> the settign is to block after 3 attempts, i simualte a wrong password but its not blocking at all. any idea |
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| 18:37.30 | [TK]D-Fender | ruben231: that is only one out of the 3 things I told you to show |
| 18:38.21 | rogers | Having trouble setting up an IAX2 peer over site to site vpn. Everything looks good in config, 1 can see the other and make calls over the IAX2, however the other side sees it as unreachable. What should I check? |
| 18:39.25 | ruben231 | <PROTECTED> |
| 18:49.10 | madduck | Do people have experience with solutions that allow for flexible, dynamic association management between incoming call sources, telephones and users? I.e. I want all our telephones (in various offices) to be registered at all times, but I want a flexible, scriptable way to associate incoming numbers with those devices depending on where users are. |
| 18:55.13 | ChannelZ-Wk | Well you could database it. |
| 18:55.54 | ChannelZ-Wk | And/or build out a FollowMe system |
| 19:00.31 | Kobaz | ruben231: you regex is wrong |
| 19:00.54 | Kobaz | *your |
| 19:01.10 | Kobaz | and you didnt paste your asterisk log |
| 19:02.15 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 19:02.16 | *** mode/#asterisk [+o file] by ChanServ |
| 19:06.13 | ruben231 | <PROTECTED> |
| 19:06.21 | ruben231 | actually - please help |
| 19:06.34 | *** join/#asterisk overfuse (~simmonds@188.29.164.231.threembb.co.uk) |
| 19:06.59 | Kobaz | it's not hard |
| 19:07.05 | Kobaz | /var/log/asterisk/full |
| 19:07.15 | ChannelZ-Wk | what log do you even have it looking at? |
| 19:08.29 | ChannelZ-Wk | and if I remember right the first bunch of your patterns (Registration from .* failed for ...) aren't really valid because the IP shown will usually be YOURS |
| 19:09.14 | overfuse | Hey, I am reading Asterisk: The Definitive Guide (4th) and I have a question... |
| 19:09.41 | ChannelZ-Wk | (but I could be wrong - I do remember than the 'fake rejection' messages didn't include the IP at all) |
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| 19:12.10 | overfuse | the book stayes, "Open up the CLI in |
| 19:12.12 | overfuse | order to see the call progression. |
| 19:12.48 | ChannelZ-Wk | asterisk -rvvv |
| 19:13.01 | overfuse | ChannelZ-Wk: Thanks |
| 19:13.27 | ChannelZ-Wk | People will also refer to it as 'the console' |
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| 19:14.17 | *** mode/#asterisk [+o file] by ChanServ |
| 19:15.15 | overfuse | ChannelZ-Wk: That was it, so far the book and only really been doing a 'asterisk -r' and had failed to mention the -rvvv. It worked perectly. Cheers |
| 19:21.59 | ruben231 | Kobaz: -------> this is the logs for my test attempt please see if thsi is ok ------> http://pastebin.com/icfaqdm4 - its not bloking i specify 4 attempts |
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| 19:23.30 | ruben231 | any idea..? please |
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| 20:18.13 | markmcn | Anyone here using queuemetrics I could do with picking some brains |
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| 20:28.07 | madduck | ChannelZ-Wk: yeah, but that would take me weeks to develop. ;) I was looking for something ready to go ;) |
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| 21:21.23 | Kobaz | ruben231: it looks like one of your regexes will match |
| 21:21.32 | Kobaz | ruben231: make sure you're checking against the correct log file |
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| 21:30.13 | Kobaz | ruben231: did you make a jail in fail2ban for asterisk? |
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| 21:36.32 | talntid | can someone do me a favor? |
| 21:36.39 | talntid | http://www.microsoftstore.com/mssg/en-SG/store/help/contact-us |
| 21:36.49 | talntid | the Microsoft Store Sales and Customer Support: phone number... call that and tell me what you get? |
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| 21:57.19 | markmcn | [TK] I recall you pointing out agents were a thing of the past as channels go but is there any way to assocaiate an agent channel with an extension without using agentcallbacklogin |
| 21:57.46 | markmcn | AgentLogin is no good as it requires the agnets to stay on the line |
| 22:00.19 | [TK]D-Fender | perhaps you should disassociate the term "agent channel" and describe the cirecumstances of the call independently |
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| 22:01.34 | markmcn | the issue is more queuemetrics rather then asterisk the problem being it's following the sip-channels in the queue log rather agents as now the sip channels are added to the queue rather then the agent-channels |
| 22:01.40 | markmcn | if that makes sense |
| 22:02.59 | [TK]D-Fender | markHow is that "bad"? |
| 22:03.47 | markmcn | becasue out call centre agents hot desk and the manager can't see what calls go to what agent they see what calls to which ext |
| 22:04.48 | [TK]D-Fender | At which point you should do what I told you earlier and use LOCAL channels as members, and not SIP devices |
| 22:06.00 | markmcn | sorry that part got lost on my earlier |
| 22:06.22 | markmcn | i'm still picking this up at a steep pace sorry to have to re-ask |
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