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| 06:14.38 | EdLin | I've just purchased a Polycom IP 430 used on eBay, will it be adequate for VoIP? |
| 06:14.59 | EdLin | I'm kind of a new VoIP person, only have used IM before. |
| 06:15.30 | EdLin | though I did set up SIP fine using Jitsi now. |
| 06:17.16 | ldc | EdLin: yeah |
| 06:18.10 | EdLin | ldc, thanks. Do you have any tips concerning one, if you're familiar with them? I'm very good at configuring both text files and menu interfaces, so that part doesn't worry me. |
| 06:19.00 | ldc | EdLin: bah I gues it will take the usual settings of any other sip phone, like where to register, username and password :) |
| 06:19.03 | EdLin | I'll be using it with a hosted PBX, my apologies for asking here, though for all I know they may be running Asterisk there too. |
| 06:19.21 | ldc | you'll probably need to set nat keepalive or something like that if you're behind nat |
| 06:19.30 | ldc | and they don't do it server side |
| 06:19.45 | EdLin | what would be a typical value for nat keepalive? |
| 06:19.49 | ldc | yes |
| 06:20.58 | EdLin | I assume you mean "yes" as in "it's a true or false value". :) |
| 06:21.09 | ldc | right :p |
| 06:21.26 | EdLin | I've downloaded the manuals already. |
| 06:21.54 | EdLin | EOL for this model will be in another couple of months, incidentally, should I be concerned about vulnerabilities if it's behind a NAT? |
| 06:22.17 | ldc | EdLin: nat is not a way of securing a network |
| 06:22.41 | EdLin | I mean a firewall.... |
| 06:23.01 | ldc | if you have a stateful firewall (e.g. iptables) you need to block new connections that are forwarded from outside to inside, but allow established and related, and new from inside to outside (again, iptables speaking :p) |
| 06:23.33 | ldc | if you have a real firewall, e.g. a zone based thing (juniper srx/ssg to name one) you need to block from untrust to trust for the services you don't want |
| 06:23.37 | EdLin | it's going to be behind a consumer router, without something as nifty as openwrt for fooling directly with iptables. |
| 06:23.44 | ldc | that as a general practice. in real life you're more or less fine with nat, but don't tell anyone |
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| 06:23.54 | EdLin | lol |
| 06:25.24 | EdLin | it's got two ethernet ports, should I put my computer with the IP phone between it and my router? |
| 06:25.43 | ldc | EdLin: that's what the port is for, yes |
| 06:26.00 | ldc | although generally you have a cdp enabled switch that turns the port into a voice/data trunk to put the phone in a vlan and the pc in another |
| 06:26.39 | EdLin | well, as you can gather, I'm not exactly in an enterprise environment. |
| 06:27.43 | EdLin | I'm going to fiddle with QoS on my router some if I have problems with it, but other than that, I hope to avoid more hardware. |
| 06:31.44 | EdLin | ldc, thanks for your help. |
| 06:33.27 | ldc | np |
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| 09:40.00 | ChannelZ | Damnit. Is RTP debug broken with PJSIP or is there some new magic way to do it? I turn it on and make a call, and it spits out like 10 packets and then quits. |
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| 13:20.36 | cervajs2 | hi, i'm converting extensions.conf to DB routing. can you help me with regexp or something which converts dialplan to single numbers like _3X0 to 310,320,330,340,... ? thanks |
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| 13:37.07 | hirogen | hi |
| 13:38.11 | hirogen | anyone use avaya 1608 IP phones, they go totally dead cant get them to switch back on - checked all of google and avaya for a fix |
| 13:38.11 | hirogen | - PoE cant be as normal power lead dont power em up but sometimes they magically come back after a few days very frustrating. seems to be a known issue. changing the ports doesnt power them up. i wonder if its some kinda battery issue? |
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| 14:36.54 | *** join/#asterisk linocisco (~linocisco@111.84.193.18) |
| 14:38.01 | linocisco | for users of 30 clients in the LAN, which server or desktop with which spec is fine in terms of brands and models (in tower type or Rack size) |
| 14:38.04 | linocisco | ? |
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| 14:44.50 | ldc | linocisco: for asterisk? |
| 14:45.13 | linocisco | ldc, yes. |
| 14:45.20 | ldc | linocisco: I guess a hp microserver will do |
| 14:46.28 | linocisco | ldc, sample model please !. is it noisy? which specs of HP microserver is recommended? |
| 14:46.55 | linocisco | ldc, if I am looking for 1U sized rack mount, which will you recommend? |
| 14:47.34 | ldc | linocisco: rack mount will be noisy if you're worried about it. plenty of options for just 30 users |
| 14:47.52 | WIMPy | http://mini-itx.com/ |
| 14:48.05 | ldc | on the refurbished some hp dl360 g5 will be cheap, around 200 eur |
| 14:49.20 | ldc | or some poweredge around that price point (1950, r200) |
| 14:49.48 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-xoamkpnzvhbnskjk) |
| 14:49.48 | ldc | ah yeah, there's also the mini itx + 1u case as WIMPy pointed out |
| 14:51.01 | mutilator | we use a lil supermicro server |
| 14:51.03 | mutilator | rack mount |
| 14:51.05 | mutilator | atom proc |
| 14:51.13 | linocisco | ldc, thanks for suggestion. we are looking for new one. and our local supplier here has only new stocks. Dell poweredge has some issues according to voip-info.org on some models. I trust HP more than Dell in terms of Desktops. I am not sure about servers. |
| 14:53.28 | mutilator | http://www.supermicro.com/products/system/1U/5017/SYS-5017A-EP.cfm |
| 14:53.31 | mutilator | something like this linocisco |
| 14:53.36 | linocisco | mutilator, supermicro is somewhat strange to me, I have never used |
| 14:54.29 | linocisco | mutilator, linocisco is meaningful. Combined power of (Linux and Cisco) |
| 14:55.07 | mutilator | k.. |
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| 15:26.27 | cervajs2 | i'm converting extensions.conf to DB routing. can you help me with regexp or something which converts dialplan to single numbers like _3X0 to 310,320,330,340,... ? |
| 15:28.30 | ldc | linocisco: combined power of two broken things then :p |
| 15:28.39 | ldc | linocisco: supermicro works fine, don't worry about it |
| 15:30.02 | linocisco | ldc, my setup would be for 30 users, and need recorded voice mail storage for minimum one month and fax server feature. which model do u recommend from supermicro? |
| 15:30.54 | ldc | linocisco: the one mutilator linked should be fine |
| 15:30.58 | ldc | although it looks a bit cpu limited to me |
| 15:32.03 | mutilator | it's more than enough for 30 users |
| 15:32.47 | mutilator | they do sell more power hungry/more powerful xeon units though |
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| 15:34.47 | mutilator | lincisco needs to invest in better routers i think |
| 15:35.22 | ldc | heh |
| 15:35.28 | ldc | or a decent OS |
| 15:35.33 | [Consultant]C-As | reloading asterisk does not enable my new ari users? how do i force ari to reload? |
| 15:35.35 | mutilator | badum pshhhhh |
| 15:36.02 | mutilator | every irc client needs http://instantrimshot.com/ built in |
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| 15:39.32 | [Consultant]C-As | nvm figuired it out |
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| 15:44.50 | cervajs2 | can i simultaneosly use extensions.conf file and static realtime extensions.conf ? i want only use realtime for [general] and [globals] sections |
| 15:56.12 | *** join/#asterisk prelude2004c (prelude200@dsl-67-55-28-3.acanac.net) |
| 15:56.44 | prelude2004c | hi everyone, can i get some help. Does anyone know why confbridge has such bad quality. Everytime i login to the bridge it sounds like people are on speaker phone and it cuts out and everything |
| 15:56.49 | prelude2004c | the server is not doing anything |
| 15:56.54 | prelude2004c | normal calls seem fine |
| 15:57.01 | prelude2004c | using asterisk v 13 |
| 15:58.02 | *** join/#asterisk theron (~theron@173-20-126-202.client.mchsi.com) |
| 15:59.54 | prelude2004c | anyone |
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| 16:44.26 | prelude2004c | can i get some help. Does anyone know why confbridge has such bad quality. Everytime i login to the bridge it sounds like people are on speaker phone and it cuts out and everything. Regular calls sound fine but its only confbrige calls that sound very bad |
| 16:49.05 | file | it will be bad if 1. The timing source in use isn't very accurate or 2. The machine can't keep up with the mixing |
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| 16:51.15 | [TK]D-Fender | Also participants who also use speakerphones could cause little cut-offs on their end that cause their mics to pick you up in double and reflect that back. |
| 16:51.16 | prelude2004c | i dont think its the machine ... but how do i know about the timing? |
| 16:51.31 | prelude2004c | none on speakphone but its not just the one guy |
| 16:51.38 | prelude2004c | its everyone.. even between two parties |
| 16:51.41 | prelude2004c | its very odd |
| 16:53.33 | [TK]D-Fender | "Everyone" as in any 2 people on different phones, etc? |
| 16:55.03 | prelude2004c | yup |
| 16:55.06 | prelude2004c | exactly |
| 16:55.25 | prelude2004c | 2 people right sends the cpu's to 40% which is still ok isn't it? |
| 16:55.28 | prelude2004c | its 20% each |
| 16:55.39 | prelude2004c | but i hear a lot of clipping and background noise |
| 16:55.43 | prelude2004c | any way to clean it up? |
| 16:55.52 | prelude2004c | like get rid of background noise |
| 16:56.17 | prelude2004c | just sounds like they are in a tin can :( |
| 16:57.25 | [TK]D-Fender | is this a dedicated server? |
| 16:57.38 | prelude2004c | it is |
| 16:57.44 | prelude2004c | and currently doing nothing |
| 16:58.18 | prelude2004c | maybe somethign with timing? |
| 16:58.30 | prelude2004c | i dont' use hardware.. so its just a base linux system with asterisk 13 |
| 16:58.38 | prelude2004c | do i need to install something that helps the timing of the system? |
| 16:59.32 | prelude2004c | i was under the impression that the timing modules were included inside asterisk already |
| 16:59.38 | prelude2004c | so i don't need much else |
| 17:00.21 | prelude2004c | i ddin't install dahdi if that means anything |
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| 17:10.27 | hirogen | <PROTECTED> |
| 17:10.29 | hirogen | - PoE cant be as normal power lead dont power em up but sometimes they magically come back after a few days very frustrating. seems to be a known issue. changing the ports doesnt power them up. i wonder if its some kinda battery issue? |
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| 17:27.48 | [Consultant]C-As | what web server is used for ari services? |
| 17:28.42 | file | Asterisk has a built-in HTTP server which is used. |
| 17:29.32 | [Consultant]C-As | how can i add allow headers to it? i need this to enable CORS Access-Control-Allow-Origin: * |
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| 17:33.07 | [Consultant]C-As | never mind i found it |
| 17:33.29 | [Consultant]C-As | <PROTECTED> |
| 17:33.44 | [Consultant]C-As | <PROTECTED> |
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| 18:29.23 | [sID] | how can I change the Remote-Party-ID? |
| 18:29.25 | [sID] | I have a standard operator Remote-Party-ID: "221234567" <sip: 221234567@XX.XX.XX.XX>; privacy = full; screen = yes. |
| 18:29.32 | [sID] | I just set it like this: |
| 18:29.33 | [sID] | EXTEN => _X., 1, SipAddHeader (P-Asserted-Identity: <sip: $ {CallerID (or)} @ xx.xx.xx.xx> \; privacy = full \; screen = yes) |
| 18:29.41 | [sID] | But unfortunately it does not work and it shows in the debug such that each set himself asterisk |
| 18:30.11 | [sID] | exten => _X.,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(ani)}@XX.XX.XX.XX>\;privacy=full\;screen=yes) |
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| 18:47.34 | volga629 | Hello Everyone, I am tying set asterisk point to edge proxy, what is recommended settings for outboundproxy in term of external ip and nat ? |
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| 19:17.11 | volga629 | <PROTECTED> |
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| 19:27.02 | [sID] | OK, |
| 19:27.07 | [sID] | ok, I did |
| 19:27.07 | [sID] | How would anyone want then you need to set the P-Asserted header-Identity:; privacy = full \; screen = yes then SIPAddHeader (Privacy: id); and Set (CALLERPRES () = prohib_passed_screen) |
| 19:27.12 | [sID] | just to be safe even as CallerID (num) anonymous <- from the small and (name) Anonymous <- large |
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| 20:23.42 | cervajs2 | i want solve this problem https://issues.asterisk.org/jira/browse/ASTERISK-22708 . but i dont have skills for it. any ideas where to get freelancer? or some bounty system? or is it possible via digium commercial support? |
| 20:25.34 | rmudgett | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties |
| 20:27.35 | cervajs2 | rmudgett: tnx |
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| 20:37.51 | hirogen | hi |
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| 20:47.09 | hirogen | anyone use avaya 1608 IP phones, they go totally dead cant get them to switch back on - checked all of google and avaya for a fix |
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| 21:14.34 | *** join/#asterisk volga629 (~bendersky@CPE4c5e0cb3c2a0-CM0c473dd46e50.cpe.net.cable.rogers.com) |
| 21:15.04 | volga629 | Hello Everyone, I am tying set asterisk point to edge proxy, what is recommended settings for outboundproxy in term of external ip and nat ? |
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| 22:44.17 | *** part/#asterisk cervajs2 (~cervenka@178.148.broadband4.iol.cz) |
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| 23:34.42 | [Consultant]C-As | when i connect to ari i keep seeing this error an it shuts down --- WARNING[24619]: res_http_websocket.c:438 ws_safe_read: Websocket seems unresponsive, disconnecting ... |
| 23:34.43 | [Consultant]C-As | <PROTECTED> |
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