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| 02:50.14 | *** join/#asterisk rexwin_ (~rexwin@116.202.96.200) |
| 02:50.33 | rexwin_ | i tried a basic asterisk setup. |
| 02:50.58 | rexwin_ | but i am unable to register any phones. |
| 02:51.00 | rexwin_ | https://wiki.asterisk.org/wiki/display/AST/Registering+Phones+to+Asterisk |
| 02:51.27 | rexwin_ | sip show peers shows nothing like that in the link. how do i check the realtime log? |
| 02:54.08 | rexwin_ | it says 2 offline. the setup is exact as in the link |
| 03:10.46 | ChannelZ | If they are offline then they haven't registed. |
| 03:11.15 | ChannelZ | Turn on some verbose. You should see *something* if they are bothering to try to register |
| 03:11.32 | rexwin_ | chan_sip.c:23173 handle_request_invite: Failed to authenticate device "Test"<sip:demo-alice@192.168.19.128>;tag=3370731e |
| 03:11.37 | ChannelZ | there you go |
| 03:12.19 | ChannelZ | you either don't have a demo-alice peer or the password is wrong or something |
| 03:23.12 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 03:39.30 | rexwin_ | ok thanks. |
| 03:39.34 | *** part/#asterisk rexwin_ (~rexwin@116.202.96.200) |
| 03:43.52 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 03:46.53 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 03:56.47 | *** join/#asterisk rexwin_ (~rexwin@116.202.96.200) |
| 03:57.01 | rexwin_ | chan_sip.c:25575 handle_request_register: Registration from '"demo-alice"<sip:demo-alice@192.168.19.128>' failed for '192.168.19.1:37982' - Device does not match ACL |
| 03:57.27 | rexwin_ | now i am getting the above message. can somebody point me how to eliminate this |
| 04:00.58 | rexwin_ | ok, i got it resolved by changing the allowed ip address as it was running under vmware. |
| 04:01.40 | rexwin_ | the user is online too. :-) |
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| 05:34.31 | WIMPy | Does anyone else have the issue that 'core restart when convenient' doesn't work on 11.8-rc1? |
| 05:34.55 | WIMPy | For me it shuts down partially and then sits there doing nothing. |
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| 06:51.53 | AlbertC | has anyone had core dump issues with bria softphone putting a call on hold? |
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| 07:53.17 | ChannelZ | WIMPy: I've had that before on several versions. Never figured out a root cause |
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| 08:23.57 | moikmellah | Anyone know if it's kosher to use libpri-1.4.14 with Asterisk 1.6.1.6? Need to upgrade libpri from 1.4.10.1, but I can't really upgrade Asterisk with it.. |
| 08:24.26 | moikmellah | DAHDI is at 2.4.1. |
| 08:29.52 | *** join/#asterisk D30 (~deo@222.127.13.226) |
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| 09:17.30 | moikmellah | Oh dear God, no. Apparently 1.4.14 doesn't play nicely with 1.4.10.1 - lots of multiframe-related errors on the upgraded PBX. Only seems to be an issue if the NET side of the PRI is 1.4.14 - if I leave the NET side at 1.4.10.1 and upgrade the CPE side to 1.4.14, everything is happy. |
| 09:17.33 | moikmellah | Weird. |
| 09:18.43 | *** part/#asterisk moikmellah (~mmiller@96-42-42-57.dhcp.ftbg.wi.charter.com) |
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| 09:29.39 | WIMPy | ChannelZ: Well, it isn't new to me, either. But now it seems to never work. |
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| 10:06.32 | podman99 | hey, not an asterisk question as the call does not appear in the CLI, however, I use polycom321 phones connected via VDSL to my remote asterisk system, however one phone keeps recieving calls from Caller 100, this call is some kind of spam or attempt at hacking/detecting a PBX, how can I stop these calls coming in |
| 10:19.23 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
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| 10:39.43 | Varazir | hello |
| 10:40.27 | Varazir | where can i read about standers asterisk supports ? |
| 10:43.09 | Varazir | do it support rfc3261? |
| 10:46.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 11:32.01 | *** join/#asterisk przemoc86 (~przemoc@unaffiliated/przemoc) |
| 11:39.42 | przemoc | hello. I have a question regarding development/release cycle for Asterisk 13 LTS. Feature freeze is exepected in July. Is it a point when the branch for 13 will be created? First release is expected in October. Will it be first the official Asterisk 13 release? Looking at 12 I've seen the branch created in late August, after feature freeze and much before first release. |
| 11:40.03 | przemoc | s/first the/the first/ |
| 11:41.10 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 11:42.46 | przemoc | Should I ask that in -dev channel perhaps? |
| 11:54.01 | *** join/#asterisk As001 (~uros@82.117.198.142) |
| 11:55.02 | As001 | Hello what is the meaning of this warning "WARNING[21873]: channel.c:3658 ast_waitfordigit_full: Unexpected control subclass '14' |
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| 12:31.45 | *** join/#asterisk giany (~giany@static.164.10.4.46.clients.your-server.de) |
| 12:31.58 | giany | hi, what are the chances for asterisk to set the same media port? |
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| 12:53.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:12.47 | *** join/#asterisk ggayan (~ggayan@200.71.197.170) |
| 13:13.17 | ggayan | hey guys, I have a funny event on my asterisk log using AGI/AMI |
| 13:13.27 | ggayan | ERROR[4617][C-0000020e]: res_agi.c:1018 get_agi_cmd: Huh? Async AGI datastore disappeared on Channel Bridge/SIP/2011-00000276<ZOMBIE>! |
| 13:13.45 | ggayan | does anyone knows why could it be happening? it isn't so frequent, but happens from time to time |
| 13:13.54 | ggayan | I'm using asterisk 11.7.0 |
| 13:17.27 | mjordan | A masquerade occurred on the channel that was in the AsyncAGI loop (which would occur if the channel was transferred out of AsyncAGI). |
| 13:17.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:18.44 | mjordan | that probably shouldn't be an ERROR. It's expected to happen if a channel gets transferred out. |
| 13:20.26 | *** join/#asterisk galba (~pif@217-162-81-21.dynamic.hispeed.ch) |
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| 13:24.02 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 13:25.15 | ggayan | mjordan thanks |
| 13:33.12 | *** join/#asterisk kippi (c2325add@gateway/web/freenode/ip.194.50.90.221) |
| 13:33.16 | kippi | hey |
| 13:33.18 | kippi | I am getting this error, the unit that is sending the call is only framing for about 1/5 seconds. Any pointers would be good.Dropping incompatible voice frame on SIP/****** of format ulaw since our native format has changed to 0x8 (alaw) |
| 13:39.25 | *** part/#asterisk giany (~giany@static.164.10.4.46.clients.your-server.de) |
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| 13:59.05 | vittorio88 | Hi guys. Anyone know what "error opening SSL certificate" can mean? I cant get Asterisk to start listening on port 5061 for tls. |
| 13:59.33 | *** join/#asterisk chazzam (~chazz@donutokyo.info) |
| 14:00.00 | Chainsaw | vittorio88: Did you apply the patches I gave you? |
| 14:12.05 | Katty | morning |
| 14:15.24 | Chainsaw | Hey Katty :) |
| 14:15.38 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:16.06 | Katty | hugs Chainsaw |
| 14:16.08 | Katty | how're you dear? |
| 14:16.15 | Chainsaw | Katty: *hug* Doing good :) |
| 14:16.20 | Katty | horay! |
| 14:25.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:25.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:27.46 | Katty | hugs sruffell |
| 14:31.54 | *** join/#asterisk Zogot (~LeonProna@095-097-219-235.static.chello.nl) |
| 14:32.00 | Zogot | ahoyhoy |
| 14:32.46 | Katty | howdy Zogot |
| 14:35.46 | *** part/#asterisk Zogot (~LeonProna@095-097-219-235.static.chello.nl) |
| 14:35.57 | *** join/#asterisk Zogot (~LeonProna@095-097-219-235.static.chello.nl) |
| 14:36.03 | Zogot | hmm, wierd bug in adium |
| 14:36.11 | Zogot | can disconnect a channel tab into nowhere |
| 14:36.11 | Katty | does it have 6 legs? |
| 14:36.22 | Katty | prods [TK]D-Fender |
| 14:36.36 | [TK]D-Fender | pokes Katty |
| 14:36.39 | Katty | morning :> |
| 14:36.44 | [TK]D-Fender | indeed |
| 14:36.52 | Katty | how's things up north? |
| 14:37.25 | [TK]D-Fender | a balmy -33 with wind-chill |
| 14:37.39 | Zogot | how far 'north' is this |
| 14:37.42 | Katty | brr :< |
| 14:38.15 | Katty | pokes eppigy |
| 14:38.20 | [TK]D-Fender | Montreal, QC |
| 14:38.22 | Katty | eppigy: how cold is it out east? |
| 14:38.39 | sruffell | waves |
| 14:38.48 | Katty | sruffell: stayin warm? |
| 14:38.59 | sruffell | I broke out my 80s sweater today in an attempt to. |
| 14:39.17 | Katty | sruffell: does it have an artari logo on it? |
| 14:39.28 | sruffell | heh..no. But that would be cool... |
| 14:39.39 | Katty | agreed. then i'd have to steal it. |
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| 14:52.26 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 14:59.39 | cusco | yellows |
| 15:01.04 | markmcn | Hi All i'm using Set(foo=${FILE(somefilename,1,1,l)}) to read the first line of a file however it put's a newline on the end of the retuened line is there a way to stop this or do I just need to use something like cut() |
| 15:01.49 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:02.26 | [TK]D-Fender | sounds like a DOS formated text file |
| 15:02.45 | markmcn | file was just created with vim |
| 15:02.55 | markmcn | on the asterisk box |
| 15:02.58 | [TK]D-Fender | hrm |
| 15:03.04 | *** join/#asterisk Gokee2 (~gokee2@24-113-140-9.wavecable.com) |
| 15:03.41 | *** join/#asterisk Master_dot_csv (~Master_do@gw1.security.web.za) |
| 15:03.56 | [TK]D-Fender | CUT might work, or perhaps trime the last char off with a sub-string |
| 15:04.22 | *** join/#asterisk mjordan (~matt@nat/digium/x-uauzctcoujxcfeyt) |
| 15:04.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:04.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:04.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:07.43 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:10.41 | markmcn | going to try cut does anyone know if I can pass it the ascii code for newline or any suggestions |
| 15:12.31 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 15:17.22 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:19.52 | cusco | so.. we have been using x-lite for some years now |
| 15:19.58 | cusco | still that old x-lite 3 what so ever |
| 15:20.14 | cusco | if we wanted to upgrade, what softphone would you folks recomment to run on windows? |
| 15:20.21 | cusco | recommend |
| 15:21.16 | [TK]D-Fender | Jitsi seems to be pretty well respected these days |
| 15:21.53 | cusco | markmcn: I would rather use ${var:0:$[${LEN(${var})}-1]} |
| 15:21.57 | cusco | or something like that |
| 15:22.05 | cusco | jitsi.. googling up |
| 15:22.23 | navaismo | also zoiper |
| 15:22.59 | markmcn | thanks cusco i'll have a go and see what I can do |
| 15:24.04 | oatha | linphone |
| 15:25.53 | Katty | has also used zoiper |
| 15:26.21 | Katty | looks into jitsi |
| 15:26.46 | Katty | oooh, an msi. that's handy |
| 15:26.57 | sp00kz | for some reason the word 'zoiper' reminds me of the old cuecats |
| 15:27.01 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 15:32.06 | kippi | if I dial out to a isdn unit i am able to push audio back, but if the unit dials into me I am just getting frame errors on the unit and static on the channel, the call is coming in via a sip trunk to our asterisk |
| 15:32.10 | *** part/#asterisk Zogot (~LeonProna@095-097-219-235.static.chello.nl) |
| 15:36.32 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 15:37.06 | JeffC_NN | pretty much all the open source softphones use sipura (as far as I know) for the audio engine, so the major differences are the graphical interfaces and how they handle signaling. |
| 15:37.07 | [TK]D-Fender | kippi: The role of the ISND unit is very unclear in your description. Please rephrase it in a more linear manner and show us the complete call attempt |
| 15:37.51 | [TK]D-Fender | ISDN* |
| 15:38.17 | JeffC_NN | (crap I meant PortAudio, not sipura) |
| 15:39.35 | kippi | [TK]D-Fender: ISDN Audio Unit -> ISDN2 -> 0203 Number (voip talk) -> voip talk to Asterisk -> Asterisk Answer and MeetMe |
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| 15:45.05 | TuomasT | Hi. Why is SRTP required for encrypted SIP voip in addition to SSL/TLS? isn't it enough to just pass all the data through the SSL/TLS tunnel? |
| 15:46.24 | WIMPy | Are you talking about SIP/TLS or are you talking about a tunnel? |
| 15:47.04 | TuomasT | SIL/TLS I believe |
| 15:47.07 | TuomasT | SIP/TLS |
| 15:47.29 | WIMPy | Well, SIP doesn't carry any audio. |
| 15:47.41 | WIMPy | That's what RTP is doing. |
| 15:48.15 | TuomasT | So we have SIP protocol to handle a call and RTP protocol to transfer audio/video? |
| 15:48.27 | WIMPy | yes |
| 15:48.32 | TuomasT | Ok, fair enough |
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| 15:52.42 | frooker | hi |
| 15:52.46 | frooker | anyone here ? |
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| 15:53.36 | frooker | i need help |
| 15:53.38 | frooker | :( |
| 15:54.13 | [TK]D-Fender | ~ask |
| 15:54.14 | 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. |
| 15:57.39 | TuomasT | If I run a SIP server that doesn't have any local phone hardware connected to it then do I need DAHDI support compiled in Asterisk? I still want to support hardware phones connecting to the server over SIP protocol |
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| 16:00.29 | [TK]D-Fender | TuomasT: DAHDI would only be required for MeetMe & Page. You could use ConfBridge instead which is superior in 11+ |
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| 16:31.41 | markmcn | cusco:Just wanted to let you know that worked a treat many thanks |
| 16:41.51 | cusco | ok |
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| 18:57.26 | roramirez | Hello. I used cel with cel_pgsql. Is possible add time using gmt 0? |
| 19:00.08 | pabelanger | anything is possible, if you understand how it works |
| 19:00.35 | pabelanger | IIRC, everything defaults to the local TZ |
| 19:00.41 | pabelanger | so, you can change the TZ on your system |
| 19:02.03 | roramirez | pabelanger: but cdr is possible set datetime on gtm 0 |
| 19:02.22 | roramirez | but using cel_pgsql nos is possible for me |
| 19:02.43 | roramirez | i am set usegmtime=yes but dont working |
| 19:03.39 | roramirez | in cel cvs insert record on gmt ' |
| 19:03.40 | roramirez | 0 |
| 19:05.32 | pabelanger | likely a feature request |
| 19:05.45 | pabelanger | I added GMT support to queues a few months ago |
| 19:05.56 | pabelanger | so, you'll likely need to do the same |
| 19:06.57 | pabelanger | You might be able to setup something in dialplan, using custom CEL |
| 19:07.11 | roramirez | ok, let me check |
| 19:07.32 | roramirez | pabelanger: how add gmt support to queues? |
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| 19:08.06 | roramirez | is possible show me something? |
| 19:08.28 | pabelanger | roramirez, http://svnview.digium.com/svn/asterisk?view=revision&revision=380209 |
| 19:09.03 | roramirez | pabelanger: thanks you |
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| 19:21.37 | cusco | question: in dialplan if I want to dial peer1, wait 5 seconds, dial peer2 (while keep ringing peer1), wait 5 seconds, dial peer3 (while keep ringing 1 and 2) ... |
| 19:21.41 | cusco | until answered |
| 19:21.45 | cusco | whats the best aproach? |
| 19:21.48 | cusco | approach? |
| 19:22.34 | cusco | have a specific queue with local/ interfaces, and have a wait(5) in peer2, and a wait(10) in peer 3? and set it to ringall ? |
| 19:23.53 | wdoekes | a queue? simply dialing Local/peer1&Local/peer2&Local/peer3 should be sufficient |
| 19:24.25 | [TK]D-Fender | multiple locals. Queue's do not tier in memebers |
| 19:24.45 | pabelanger | multi-queues will |
| 19:25.04 | [TK]D-Fender | paAs in? |
| 19:25.09 | [TK]D-Fender | pabelanger: As in? |
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| 19:26.01 | pabelanger | I miss read |
| 19:26.28 | pabelanger | in the case above, you could try ringall queue, then dynamically increment weight |
| 19:27.07 | [TK]D-Fender | pabelanger: Problem is that will always break the "continue ringing". It reevaulates at each timeout and redials |
| 19:27.27 | pabelanger | well, we get around it by playing ringback as the MOH |
| 19:27.32 | pabelanger | the caller doesn't heard the break |
| 19:27.37 | pabelanger | and the logic is allowed to work |
| 19:27.42 | [TK]D-Fender | Iso it will do 1,2 then stop, then 1,2,3,4. But 1 & 2 will not be "continuous: |
| 19:27.49 | pabelanger | however, it messes with your stats |
| 19:27.53 | pabelanger | but if you don't care... |
| 19:28.11 | [TK]D-Fender | And can cause screwups when they try picking up between dials |
| 19:28.13 | cusco | we have a specific MOH for this too |
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| 19:28.26 | cusco | right now we're dialing one peer after another |
| 19:28.58 | cusco | but I would like to keep dialing the previous ones,so I could go as wdoekes said: dial(peer1,5); dial(peer1/peer2,5) so on |
| 19:29.02 | cusco | or a queue? |
| 19:29.49 | cusco | multiple locals |
| 19:29.53 | pabelanger | right |
| 19:30.02 | cusco | that means different locals interfaces, for the same queue? |
| 19:30.11 | [TK]D-Fender | cusco: Multiple Locals |
| 19:30.14 | pabelanger | IMO, local channels is the uglier approach, but you have options |
| 19:30.19 | cusco | googling up multiple locals |
| 19:30.21 | skrusty | evening |
| 19:30.38 | [TK]D-Fender | nothing to google |
| 19:30.39 | [TK]D-Fender | [14:23]wdoekesa queue? simply dialing Local/peer1&Local/peer2&Local/peer3 should be sufficient |
| 19:30.45 | cusco | ah! |
| 19:30.52 | cusco | dial peer&peer |
| 19:30.58 | cusco | ok |
| 19:31.07 | cusco | and in each local, add a delay? |
| 19:31.19 | [TK]D-Fender | of course his use of "peer1" etc is misleading and debateably inaccurate |
| 19:31.20 | cusco | right ok thanks sorry for troubling you with boring questions |
| 19:31.32 | cusco | :) |
| 19:31.48 | [TK]D-Fender | Actually, you would add the delay before the dial of those to be "added" in series |
| 19:31.58 | cusco | hu? |
| 19:32.04 | cusco | howso? |
| 19:32.06 | [TK]D-Fender | Not a boring question... it's a specific implementation |
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| 19:32.58 | [TK]D-Fender | Dial(Local/100@mycontext/n,Local/200@mycontext/n,Local/300@mycontext/n,120) |
| 19:33.06 | paulc | Like.. "Call my desk phone, but if I don't answer in 10 seconds, call my cell.. but leave the desk phone still ringing" - that kind of thing? |
| 19:33.09 | [TK]D-Fender | 100 could jut dial right away. |
| 19:33.22 | cusco | yes.. |
| 19:33.25 | cusco | and 200? |
| 19:33.34 | [TK]D-Fender | 200 would do a Wait(10) to wait 10 seconds, and THEN you'd dial them, etc |
| 19:33.38 | cusco | ah yes |
| 19:33.44 | cusco | that is what I was thinking |
| 19:33.53 | [TK]D-Fender | You don't need to wait AFTER.. there is no point in "after" |
| 19:34.24 | cusco | yes yes, my point was to wait in each specific local before the dial |
| 19:34.28 | [TK]D-Fender | Each of these get "dialed" immediately... its just that you are introducing delays before they start doing the actual dirty-work |
| 19:34.29 | cusco | thank you for your input |
| 19:35.19 | [TK]D-Fender | I introduced the 120 there as the "master timeout" to safeguard against programming errors that might let it go on indefinitely |
| 19:35.38 | [TK]D-Fender | Always consider what each leg might actually do. |
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| 20:34.22 | TuomasT | How do I verify a connection uses SRTP ? Let's say when I am calling from a client to asterisk voicemail |
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| 20:39.26 | TuomasT | Ok by using tethereal |
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| 22:24.06 | TuomasT | How to configure so that unknown clients from internet are able to connect to Asterisk AND ONLY make calls to a few defined local sips? |
| 22:28.03 | [TK]D-Fender | contexts <- |
| 22:28.32 | [TK]D-Fender | And the do not call "to sips". They dial EXTENSIOSN. What your dialplan allows... is what contexts are there to separate |
| 22:28.36 | TuomasT | So in my [guests] context I place only one local SIP so thosecan only call that? |
| 22:29.57 | TuomasT | replace "local SIP" with "one extensions" |
| 22:30.03 | TuomasT | one extension |
| 22:31.03 | [TK]D-Fender | You determine what any give class of caller gets to do |
| 22:36.19 | TuomasT | Hmm. I have sip.conf: [general] allowguest=yes context=guests and extensions.conf: [guests] exten => XXXX,1,Dial(SIP/XXXX). But I can't call that SIP (=XXXX@myip) ) with linphone without authentication request popping up |
| 22:36.51 | markmcn | Is it possible to prefix CID on an inbound route rather then when the call enters a queue? |
| 22:37.24 | [TK]D-Fender | markmcn: You can do whatever you want before calling Queue() |
| 22:37.44 | markmcn | Nice one thx |
| 22:37.55 | [TK]D-Fender | TuomasT: * will still try to identify the calleer... and will then let them through if they don't think they know who it is, or did and succeeded |
| 22:38.09 | [TK]D-Fender | TuomasT: So go look at that call. REALLY close |
| 22:39.26 | ChannelZ-Wk | And then beat it like it owes you money |
| 22:40.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.248) |
| 22:41.12 | TuomasT | This is just testing so the worst things that could happen now is someone breaks into shell access through bug asterisk |
| 22:41.19 | TuomasT | bug in |
| 22:41.46 | TuomasT | [TK]D-Fender: "* will still try to identify the calleer" huh ? |
| 22:41.53 | [TK]D-Fender | TuomasT: And how many times has that happened in the past? |
| 22:42.04 | TuomasT | [TK]D-Fender: for me, 0 |
| 22:42.10 | [TK]D-Fender | and everyone else? |
| 22:42.19 | TuomasT | unknown |
| 22:42.24 | [TK]D-Fender | You stated a feared scenario .... not what is the realistic odds of that? |
| 22:42.37 | [TK]D-Fender | We know... that's why security notices are posted |
| 22:43.16 | [TK]D-Fender | If you want to avoid a shell-access risk .... block AMI. |
| 22:43.21 | [TK]D-Fender | tahts threat #1 |
| 22:43.25 | TuomasT | AMI? |
| 22:43.41 | [TK]D-Fender | ~book |
| 22:43.41 | 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 |
| 22:43.43 | [TK]D-Fender | ^^^ |
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| 22:44.04 | markmcn | Great book helped me a fair bit this week |
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| 22:45.52 | TuomasT | Ok one important security question: does asterisk disable everything unless enabled by default including AMI? |
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| 22:46.10 | [TK]D-Fender | There is no such thing as "default". |
| 22:46.16 | [TK]D-Fender | There is only what YOUR configs tell it to. |
| 22:46.39 | TuomasT | make samples configs |
| 22:46.51 | [TK]D-Fender | Then you have what the samples give you. Time to get reading.... |
| 22:47.32 | TuomasT | There are 104 config files |
| 22:48.16 | TuomasT | I can't verify all of them especially with my limited asterisk knowledge |
| 22:48.31 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/ |
| 22:48.35 | [TK]D-Fender | Doesn't look like 100 |
| 22:49.01 | TuomasT | ls *.conf -l | wc -l 98 |
| 22:49.02 | [TK]D-Fender | hrm... actually.. the count DOES. |
| 22:50.10 | mjordan | TuomasT: core show settings will provide you some of what is enabled/disabled. |
| 22:50.58 | mjordan | TuomasT: From a security PoV, you may also want to read the README.-SERIOUSLY.bestpractices.txt file included with Asterisk |
| 22:52.43 | mjordan | TuomasT: if you aren't using AMI, then your callers only have the power you give them in the dialplan. Remote execution of scripts, etc. are only going to occur if a caller executes that dialplan application. They don't have the ability to add that dialplan application themselves. |
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| 23:01.37 | TuomasT | mjordan: these were good tips/advices |
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| 23:17.31 | TuomasT | If I call sip:XXXX@yyyy.com then will asterisk interpret the XXXX part as an extension? |
| 23:17.42 | TuomasT | to call to |
| 23:20.29 | TuomasT | or how else would I call extension XXXX at sip server yyyy.com |
| 23:20.40 | *** part/#asterisk mjordan (~matt@nat/digium/x-uauzctcoujxcfeyt) |
| 23:25.43 | JeffC_NN | TuomasT: Yes. This explains SIP URIs. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ICR-SECT-1.html |
| 23:31.31 | TuomasT | Shouldn't this allow [guests] to call extension 2: extensions.conf: [guests] exten => 2,1,Dial(SIP/2). The asterisk server responds by 401 unauthorized when outside calls 2@yyyy.com with that conf |
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| 23:46.35 | JeffC_NN | TuomasT: Have you set allowguest=yes in sip.conf? Did you reload sip afterward? |
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| 23:53.49 | TuomasT | JeffC_NN: Yes. Still can't get it to work. Server always requires authentication after the client send INVITE |
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| 23:55.12 | TuomasT | The server does respond with "OK" when the client first sends OPTIONS |
| 23:55.35 | TuomasT | if the supplied extension matches the one I allow guests to call to |