| 00:22.47 | *** part/#asterisk kharwell (kharwell@nat/digium/x-vmbbdtjczosjefsy) |
| 00:23.39 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 00:55.36 | *** join/#asterisk kritzikratzi (~kritzikra@80.120.104.90) |
| 01:00.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:02.08 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 02:06.31 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 02:18.25 | *** join/#asterisk klow (~klong@c-67-161-106-89.hsd1.wa.comcast.net) |
| 02:19.20 | klow | <PROTECTED> |
| 02:19.49 | klow | ^ does that mean asterisk/pjsip is actually trying to make an outbound TLS connection to my SIP phone ? |
| 02:21.54 | *** join/#asterisk ttaylor (~ttaylor@vpn.duh.net) |
| 02:22.55 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 02:45.49 | klow | Got 2 soft phone which can register to Asterisk, and a simple dial plan which I thought would work, but when I dial one from the other it does not ring and eventually times out . Whole slew of logging but I don't see anything obvious to my eyes |
| 02:46.46 | klow | anyone able to have a look at the log? thanks http://pastebin.com/iEjb6iGG |
| 02:48.00 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 03:20.04 | carrar | man busy day |
| 03:20.07 | carrar | but now done |
| 03:20.24 | carrar | time 4 lunch! |
| 03:20.38 | carrar | and wrong channel, NICE |
| 03:23.59 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-mcjopujpvzzaquwo) |
| 03:41.49 | *** join/#asterisk F2Knight_ (~F2Knight@morgansackett.snozzages.com) |
| 03:42.24 | drmessano | You hate us |
| 03:47.16 | carrar | haha |
| 03:47.21 | carrar | https://www.kickstarter.com/projects/pine64/pine-a64-first-15-64-bit-single-board-super-comput |
| 03:47.26 | carrar | hows that for sharing hte love |
| 04:12.42 | *** join/#asterisk doome_ (~doome@94-21-194-74.pool.digikabel.hu) |
| 04:17.55 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 04:54.55 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 10:10.20 | *** join/#asterisk infobot (ibot@rikers.org) |
| 10:10.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.0 (2016/01/15), 11.21.0 (2016/01/15); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 10:32.21 | *** join/#asterisk Bordr (~Bordr@23-24-156-49-static.hfc.comcastbusiness.net) |
| 10:39.30 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 10:51.38 | *** join/#asterisk todwetsprock (~florian@p5099a594.dip0.t-ipconnect.de) |
| 10:54.17 | *** join/#asterisk rsanting (~rsanting@217.21.195.34) |
| 11:07.34 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 11:19.11 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 11:29.40 | *** join/#asterisk mccar (~chatzilla@195.208.45.210) |
| 11:31.15 | mccar | Hi! Can anybody advice good article about many peers on one IP? Or maybe anyone did multiple VoIP connections from GOIP8 to asterisk? |
| 11:39.06 | bl4ckh4t | <PROTECTED> |
| 11:39.23 | bl4ckh4t | Where are you bassed? |
| 11:39.47 | mccar | Russia |
| 11:42.17 | mccar | I need to configure GOIP8... It has 8 SIM and calls to asterisk well, but CID selected randomly and I cant figure out on which SIM I have a call |
| 11:49.40 | *** join/#asterisk keithf (~keithf@ool-2f151dc2.static.optonline.net) |
| 12:04.55 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:14.30 | *** join/#asterisk kritzikratzi (~kritzikra@80.120.104.90) |
| 12:33.44 | *** part/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 12:44.24 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 12:46.17 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 12:46.33 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 12:48.51 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 12:54.36 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 12:59.27 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:13.01 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 13:26.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:36.16 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 13:38.31 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:38.40 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 13:39.53 | yang | What is the quality of Aastra/Mitel VoIP phones ? Would you recommend them for purchase or rather some other brand ? For SIP phones ? |
| 13:40.46 | *** join/#asterisk anonymouz666 (bd192dc9@gateway/web/freenode/ip.189.25.45.201) |
| 13:42.06 | [TK]D-Fender | Generally good. Choosing them depends on the model & reason |
| 13:42.16 | [TK]D-Fender | For general use Polycom > All |
| 13:42.23 | yang | well just for making occasional calls, not more than 5 calls/day |
| 13:42.28 | yang | ok |
| 13:42.29 | [TK]D-Fender | But there are a few applications I like Aastra for |
| 13:43.09 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:43.29 | [TK]D-Fender | I use Aastra 6739i's for my CSR's because of the on-screen colour BLFs |
| 13:43.49 | [TK]D-Fender | so they know which of the other agents is on the phone clearly |
| 13:44.00 | [TK]D-Fender | And is a more readable screen |
| 13:44.16 | yang | So you'd recommend Polycom over Aastra still, may I ask why ? |
| 13:44.25 | yang | i do have a Polycom phone which is fine |
| 13:45.01 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ycmuyafbrhhcnkfz) |
| 13:45.01 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:45.13 | [TK]D-Fender | physical build & audio quality |
| 13:45.49 | [TK]D-Fender | most of the Aastras had shit rubbery buttons, handset with no weight to it and the speaker & speakerphone were a little tinny |
| 13:45.57 | yang | ok |
| 13:46.24 | [TK]D-Fender | Polycom's adhoc call handling is still top of class |
| 13:46.50 | WIMPy | What's that? |
| 13:48.40 | yang | thanks for the hints [TK]D-Fender |
| 13:48.53 | [TK]D-Fender | just the ability to have mulitiple line-keys with possibly multiple calls each, and take multiple and join them in a conference without having to expressly call out to #3 to bring them in |
| 13:49.05 | [TK]D-Fender | on a call and another comes in? 3-way. |
| 13:49.31 | [TK]D-Fender | Or call out for it. Join & split whenever. Also when you hang up it'll rebridge them so they can keep talking. |
| 13:49.34 | [TK]D-Fender | Not all phones do this |
| 13:50.01 | WIMPy | Multiple buttons with multiple calls each sounds like what I think of the biggest issue with many SIP phones. |
| 13:50.55 | WIMPy | And I still think that doing conferences on a terminal isn't exactely the cleverest idea. |
| 13:51.12 | *** join/#asterisk theron (~theron@2620:10d:c091:200::c:692e) |
| 13:51.30 | anonymouz666 | Lots of good videos on youtube asterisk channel |
| 13:51.32 | anonymouz666 | Astricon 2015 |
| 13:52.16 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-drghqskqmhmcnuft) |
| 13:56.37 | newtonr | anonymouz666, yes! I agree |
| 13:57.20 | newtonr | I also recommend mjordan's series of recent blog posts: https://www.asterisk-blog.com/2016/01/04/how-to-contribute-to-asterisk-part-one/ |
| 13:57.38 | mbecroft | [TK]D-Fender: are you saying the polycom phones support >3 way conferencing with all features on the phone, without need for asterisk support? |
| 13:59.45 | mbecroft | I have a client where very easy to use conferencing support is #1 priority. Having it supported in the phone UI, without having to call into conference rooms and the like would be pretty good |
| 14:01.51 | robmal | Which polycoms? |
| 14:05.35 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 14:10.12 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:18.29 | *** join/#asterisk kritzikratzi (~kritzikra@80.120.104.90) |
| 14:27.35 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 14:29.30 | mbecroft | robmal: I don't know; the ones [TK]D-Fender was referring to |
| 14:32.37 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 14:42.59 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 14:49.31 | [TK]D-Fender | mbecroft: 5-way IIRC |
| 14:49.40 | [TK]D-Fender | mbecroft: In newer firmwares/models |
| 14:49.48 | [TK]D-Fender | mbecroft: Which I don't have at my office |
| 14:50.04 | robmal | 4 |
| 14:51.19 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:51.42 | anonymouz666 | anyone already used res_xmpp to get outside devicestate info? |
| 14:52.26 | anonymouz666 | i.e.: get skype for business XMPP presence state and input into asterisk |
| 14:52.46 | anonymouz666 | I am big user of corosync, but in this case it's not possible |
| 14:56.38 | anonymouz666 | the XMPP solution is not a good way either |
| 14:57.31 | anonymouz666 | res_pjsip is a much more reliable solution |
| 14:58.44 | *** join/#asterisk kharwell (kharwell@nat/digium/x-psuwqwivafwmbeod) |
| 15:08.00 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 15:16.24 | *** join/#asterisk captain118 (~Adium@142.sub-70-193-111.myvzw.com) |
| 15:18.53 | *** join/#asterisk F2Knight (~F2Knight@morgansackett.snozzages.com) |
| 15:19.41 | *** join/#asterisk youjelly (~youjelly@75.141.246.18) |
| 15:19.45 | youjelly | what would be the best provider to get a DID for Kingdom of Saudi Arabia? |
| 15:20.57 | *** join/#asterisk F2Knight (~F2Knight@morgansackett.snozzages.com) |
| 15:34.22 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:34.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:36.35 | *** part/#asterisk lumasepa (~gestoip@193.145.124.30.local.ull.es) |
| 15:37.31 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-tbewltxpxrzlvpsj) |
| 15:37.43 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-tzafftwynpwjcroy) |
| 15:39.14 | *** join/#asterisk theron (~theron@2620:10d:c091:200::c:db56) |
| 15:54.17 | *** join/#asterisk mjordan (mjordan@nat/digium/x-borepgpnlnlzllpp) |
| 15:54.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:56.55 | cusco | hi folks |
| 15:57.17 | cusco | using realtime, sql configuration, wrapuptime for queues, is set in seconds or milliseconds ? |
| 15:59.42 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 16:02.10 | *** join/#asterisk kharwell (kharwell@nat/digium/x-gpsndrfnsdptdfgn) |
| 16:03.59 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/queues.conf.sample?revision=430127&view=markup |
| 16:04.04 | [TK]D-Fender | ^ |
| 16:08.07 | anonymouz666 | [TK]D-Fender: already using res_pjsip? |
| 16:08.16 | [TK]D-Fender | Not yet. Soon |
| 16:08.40 | *** part/#asterisk kritzikratzi (~kritzikra@80.120.104.90) |
| 16:08.51 | anonymouz666 | Did you see the astricon 2015 video? moving from chan_sip to res_pjsip? |
| 16:09.42 | [TK]D-Fender | nope |
| 16:12.20 | anonymouz666 | could make things more clearer |
| 16:16.32 | WIMPy | wonders if he really wants to know when a registration expires for a peer that was unreachable already. |
| 16:19.34 | anonymouz666 | where is it? |
| 16:21.34 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ippzqptgndujhesu) |
| 16:41.40 | *** join/#asterisk bramgn (~bram@2a02:e00:fffb:109::) |
| 17:00.03 | anonymouz666 | can I use the latest pjsip version 2.4.3 or should I keep stuck in previous version? |
| 17:00.59 | file | 2.4.3 is fine |
| 17:08.48 | *** join/#asterisk kharwell (kharwell@nat/digium/x-fzprsfrheulsmdbl) |
| 17:31.50 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-zfvfakvjgytwwjci) |
| 17:31.56 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:36.18 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 18:04.32 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 18:28.24 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 18:48.06 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 18:48.25 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 18:53.30 | drab | Hi, i' |
| 18:54.01 | drab | Hi, i'm on asterisk 11, trying to setup the manager, very basic test, here's the config: http://pastebin.com/S3MJXGmL |
| 18:54.33 | drab | I'm trying to use the AMI for monitoring with nagios. I want only the nagios host to be able to talk to the AMI |
| 18:55.25 | drab | I thought what I put in the config would have achieved that, but it doesn't, I can connect to the AMI from my workstation too (diff ip, the nagios host is on its own system on .5) |
| 18:55.31 | drab | what am I doing wrong? |
| 18:56.05 | drab | my understanding of deny/permit is that it'd blocked every ip and then only allow what's in the permit line |
| 18:56.32 | robmal | manager reload |
| 18:56.56 | drab | already done and I now it did it also because I had the permissions wrong and my test commands were rejected |
| 19:01.03 | drab | just re-did it for good measure, same thing, I can still connect from my workstation |
| 19:22.20 | *** join/#asterisk josefig (~josefig@unaffiliated/josefig) |
| 19:26.15 | *** join/#asterisk newtonr (~RustyNewt@173-21-146-94.client.mchsi.com) |
| 19:26.15 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:30.03 | *** join/#asterisk Demon_VoIP (~demon@ip253.net222.n37.ru) |
| 19:52.31 | *** join/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 19:55.21 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 19:57.04 | *** part/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 20:24.50 | klow | Looking for some help in diagnosing why a simple pjsip 2 extension setup - my soft phones register and I can call a dummy extension and playback a sound but calling phone->phone doesn't ring the phones |
| 20:25.38 | klow | http://pastebin.com/iEjb6iGG |
| 20:25.49 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 20:26.57 | newtonr | klow, you don't need debug on for that. turn on warning, error, notice, verbose, and "pjsip set logger on" |
| 20:27.06 | newtonr | turn off debug and get a new log |
| 20:28.49 | newtonr | but it looks like the endpoint dialed was busy. |
| 20:28.58 | newtonr | the other log would reveal more |
| 20:30.03 | klow | ok |
| 20:34.44 | newtonr | dialstatus was CHANUNAVAIL - so probably not registered or something along those lines |
| 20:41.06 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 20:47.08 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 21:01.16 | *** join/#asterisk K0HAX (~K0HAX@shellhost.home.englehorn.com) |
| 21:03.04 | *** join/#asterisk klemorali (~klemorali@71-85-70-54.static.spbg.sc.charter.com) |
| 21:08.32 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 21:20.03 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 21:50.45 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:59.04 | drab | trying my luck again, still having issues with AMI allowing connections from ip that should in fact be denied: here's the config: http://pastebin.com/S3MJXGmL |
| 21:59.44 | drab | I've done all the debugging I could think of and can't figure out why connections from my workstation (ie not the nagios host) are allowed |
| 21:59.58 | [TK]D-Fender | Show us |
| 22:01.03 | drab | show us what? I can telnet in from my workstation, use the nagios credentials, and I successfully log in and can run SIPpeerstatus |
| 22:01.15 | drab | want me to paste the successful login? |
| 22:01.23 | drab | and the ifconfig |
| 22:01.38 | drab | to prove it's a diff host? i'm seeing the logins/logoff from the asterisk cli too, can paste those too |
| 22:02.50 | [TK]D-Fender | ALL of it |
| 22:03.56 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fzprsfrheulsmdbl) |
| 22:05.54 | *** join/#asterisk captain118 (~Adium@50-81-28-72.client.mchsi.com) |
| 22:08.29 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 22:11.36 | [TK]D-Fender | permit=192.168.0.5/255.255.255.0 |
| 22:11.42 | [TK]D-Fender | Mind you taht is a total CIDR fail |
| 22:11.59 | [TK]D-Fender | you have a MASK against the last byte and are allowing your entire SUBNET |
| 22:12.07 | [TK]D-Fender | That is not the mask of a HOST |
| 22:20.04 | drab | that was a total CIDR fail indeed, thanks, I just typed ip/netmask of the host I wanted to allow |
| 22:20.10 | drab | that was it, works now, thank you |
| 22:20.42 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 22:25.07 | klow | Question, when you do "sip show peer" it shows an IP address for the peer. Anyone know where this actually is populated from? Is this the IP the server will reach out to to get a call , or to send RTP data to? My issue is that some of my soft phone clients connect to a VPN from outside, so they are VPN'd to the LAN of my asterisk server. Their peer address in "sip show peer" is showing the LAN ip of my firewall, instead of the IP of t |
| 22:25.08 | klow | heir VPN client which is what would actually be reachable. I don't know why this ip is showing in "sip show peer" , because testing for example a netcap tcp connection from a vpn client, the client IP shows up correctly |
| 22:27.59 | klow | and my clients register with tis which would be a tcp connection , so asterisk should see the correct client IP in the registry i would thing. |
| 22:28.01 | klow | think* |
| 22:28.05 | newtonr | IIRC that should be the IP the SIP client tells Asterisk to contact it at |
| 22:28.30 | newtonr | What version of Asterisk are you using? is this a new install? |
| 22:30.03 | newtonr | If you are learning Asterisk you definitely don't want to get started with chan_sip. You'll want to start learning chan_pjsip https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 22:30.23 | [TK]D-Fender | * will send to whatever IP the devcice tells it when called unless told to ignore it |
| 22:30.28 | klow | thanks newton so this is a "production" freepbx system with asterisk 11 |
| 22:30.37 | klow | i am doing some asterisk 13 from scratch w/ pjsip |
| 22:30.42 | klow | in hopes to replace this system |
| 22:30.43 | newtonr | cool cool |
| 22:30.57 | klow | but .. for now just troubleshooting these vpn problems. ppl want the vpn on, for the security, blah blah |
| 22:31.22 | *** join/#asterisk poolson (~dddd@c-98-247-198-255.hsd1.wa.comcast.net) |
| 22:31.29 | klow | i see chan sip has the incorrect IP from the get go with the registry |
| 22:31.32 | klow | <PROTECTED> |
| 22:31.35 | klow | ^ wrong IP . |
| 22:32.05 | poolson | hey when using DPMA is there any caveats to doing custom apps ? i can load one directly to the phoen from my computer .. but not from asterisk |
| 22:32.20 | poolson | is there something else that needs to be happen before it gets pushed to device ? |
| 22:32.28 | klow | so I though it was a bad NAT issue , but my tcp netcat on the vpn client and then listener on the server shows correct client IP |
| 22:33.36 | drab | is there a preferred python library to talk to the AMI to monitor peer status? I'm seeing a bunch of stuff that seems mostly deprecated |
| 22:34.10 | drab | python-asterisk, pyst, pyst2 (forked beacuse pyst's maintaner apparently wasn't responding), starpy but that's just twisted |
| 22:34.24 | bramgn | newtonr: out of curiosity, what's wrong with learning asterisk with chan_sip instead of pjsip? I haven't heard of that before (and i'm just starting to play with asterisk) |
| 22:34.31 | drab | almost tempted to just open a socket and read stuff, but wondering what people are doing here |
| 22:34.36 | [TK]D-Fender | klow, that is where the CLIENT told * to send calls to unless * was told to IGNORE what the client offered |
| 22:35.13 | klow | got it |
| 22:35.18 | klow | OK ! thanks guys |
| 22:35.27 | newtonr | bramgn, http://blogs.digium.com/2013/11/20/asterisk-12-part-iv-sip-stack-future/ |
| 22:36.04 | bramgn | newtonr: thanks, i'll dive right in :) |
| 22:36.35 | newtonr | bramgn, in short, since Asterisk 12, res_pjsip/chan_pjsip is the focus for SIP support in Asterisk. |
| 22:37.04 | newtonr | bramgn, that blog article explains some motivation - here is the latest docs: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 22:37.23 | bramgn | then it would indeed not make much sense to start learning chan_sip if that's on its way out |
| 22:38.05 | newtonr | bramgn, yeah :) |
| 22:39.24 | drab | also with the AMI read/write settings, is there any way to limit a user to a specific call? I don't want this nagios thing to be able to send any other command or do anything except execute SIPpeerstatus |
| 22:39.43 | drab | but it seems at most what I can do is to grant system,log,whatever privileges which grant all kinds of access |
| 22:39.54 | drab | don't see a way to only allow a specific command |
| 22:40.02 | drab | am I missing something? |
| 22:40.17 | drab | this on Asterisk 11 |
| 22:41.20 | *** join/#asterisk captain118 (~Adium@50-81-28-72.client.mchsi.com) |
| 22:45.54 | [TK]D-Fender | <drab> also with the AMI read/write settings, is there any way to limit a user to a specific call? I don't want this nagios thing to be able to send any other command or do anything except execute SIPpeerstatus <- no |
| 22:46.35 | [TK]D-Fender | Create your own proxy if you want something like this |
| 22:55.27 | *** join/#asterisk sgriepentrog_ (a2ef2342@gateway/web/freenode/ip.162.239.35.66) |
| 22:58.54 | sgriepentrog_ | head somebody was asking about dpma? |
| 23:04.25 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 23:11.11 | klow | Besides full-blown PBX management such as freepbx, are there any good tools to visualize asterisk , calls reports, system performance, etc ? |
| 23:11.22 | klow | open source ones i mean .. |
| 23:14.15 | [TK]D-Fender | Plenty of call reporting apps out there |
| 23:16.26 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |