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| 09:24.22 | Jeka___ | hi, cant change language. whats problem? |
| 09:24.51 | Jeka___ | [Nov 28 18:57:15] ERROR[13801]: sounds_index.c:86 get_languages: Failed to stat /var/lib/asterisk/sounds/ru |
| 09:25.12 | Jeka___ | wget + tar + mv |
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| 09:25.44 | Jeka___ | pbx*CLI> core show settings PBX Core settings ----------------- Version: 13.0.1 |
| 09:25.55 | Jeka___ | Entity ID: 00:80:48:12:ef:89 Default language: ru Language prefix: Enabled |
| 09:26.26 | Jeka___ | centos 6.6, from packages.asterisk.org |
| 09:27.32 | Jeka___ | 00085 if (stat(ast_str_buffer(variant_dir), &st) < 0) { 00086 ast_log(LOG_ERROR, "Failed to stat %s\n", ast_str_buffer(variant_dir)); 00087 continue; 00088 } 00089 |
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| 09:29.04 | Jeka___ | stat /var/lib/asterisk/sounds/ru |
| 09:29.08 | Jeka___ | <PROTECTED> |
| 09:29.16 | Jeka___ | <PROTECTED> |
| 09:30.30 | Jeka___ | sounds_index.c:86 get_languages: Failed to stat - whats up ??? |
| 09:32.23 | Jeka___ | Oo anybody alive? |
| 09:43.07 | Jeka___ | if (stat(ast_str_buffer(variant_dir), &st) < 0) - then ERROR got error num, whats error may be here? |
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| 10:27.56 | coreyfarrell | check permissions on the dir you extracted, probably owned by root and user asterisk cannot access |
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| 11:43.26 | Greenlight | Morning folks, been a while |
| 11:43.42 | Chainsaw | Greenlight: Certainly has been. Welcome back :) |
| 11:43.48 | Greenlight | ty :) |
| 11:45.42 | Greenlight | IAX2 trunking seems so great on paper, and even in testing, but in practice it seems to just give problems with call quality. Anyone got some pointers on using it on low bandwidth links? |
| 11:48.25 | Greenlight | Squeezing 100 calls over a meg reliably would be so awesome |
| 11:48.58 | Chainsaw | Greenlight: I think jkroon would be a good one to talk to. |
| 11:49.11 | Chainsaw | Greenlight: Just appeared online in #gentoo-voip |
| 11:49.54 | Greenlight | Great idea! |
| 11:52.59 | Chainsaw | Greenlight: It's probably this one: https://issues.asterisk.org/jira/browse/ASTERISK-19521 |
| 11:53.53 | Chainsaw | Greenlight: Hrm, no. That's closed. |
| 11:54.37 | Greenlight | Indeed |
| 11:54.57 | Chainsaw | Greenlight: Oddly that patch still applies and is still in our patch set. |
| 11:55.42 | Chainsaw | Greenlight: My patch explanatory text contains: "This builds on the divisor fix by seanbright, but both are required to fully resolve audible artifacts." |
| 11:55.45 | Chainsaw | Emphasis mine. |
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| 11:56.18 | Greenlight | That seems plausible then |
| 11:56.41 | Chainsaw | Greenlight: So if that works for you, please consider re-opening that report or filing a new one. |
| 11:57.51 | Greenlight | Just reading over it now |
| 11:59.03 | Chainsaw | Greenlight: http://distfiles.gentoo.org/distfiles/gentoo-asterisk-patchset-3.13.tar.bz2 |
| 11:59.18 | Chainsaw | Greenlight: 07-iax2-honour-trunk-frequency.patch |
| 11:59.51 | Greenlight | Why could the bug show as closed if it was never fixed and the patch never commited? Odd isn't it? |
| 12:00.10 | Chainsaw | Greenlight: I cannot think of a diplomatic way to phrase this, so will remain silent on the matter. |
| 12:01.22 | Greenlight | Ahh... one of those! |
| 12:01.30 | Chainsaw | Greenlight: There's an unprotected access fix as well, but that would crash. |
| 12:06.24 | file | according to the log it was committed |
| 12:08.11 | Chainsaw | file: But if it was committed as on that report, my downstream patch shouldn't apply. |
| 12:08.36 | file | I can't apply it to my 1.8 or trunk working copies |
| 12:09.21 | Chainsaw | file: How about Asterisk 11? |
| 12:09.41 | file | nope |
| 12:12.29 | file | I've also confirmed, by looking at the commit and the code, it's in there |
| 12:13.04 | Chainsaw | file: Then I must have a faulty rediff, let me check and make sure it disappears in that case. |
| 12:13.25 | file | if you search for "trunkmtu" in chan_iax2.c you can find the area |
| 12:14.35 | Chainsaw | file: Yes, let me do a test merge and see what I end up doing. It looks like I'm adding a second one that is never used. |
| 12:15.58 | file | nods |
| 12:17.00 | Chainsaw | file: Very silly. There's two of them. |
| 12:17.15 | Chainsaw | Greenlight: You have patch 07 once, which should be ample. |
| 12:17.31 | Chainsaw | Greenlight: So it's probably about specifying trunkfreq like jkroon does. |
| 12:17.48 | Chainsaw | would like to state that Digium is blameless and that they were right to close the report |
| 12:18.36 | file | it probably closed automatically as a result of the commit going in |
| 12:18.51 | file | in the future if you click the "Source" tab you can see where it all goes |
| 12:18.59 | Chainsaw | nods |
| 12:19.18 | Greenlight | Cool, lets hope that the trunkfreq is to blame |
| 12:19.38 | Greenlight | And it's not some timing related issue or other such headache |
| 12:19.52 | Greenlight | IAX2 trunking would be so awesome if it works well |
| 12:20.38 | Chainsaw | Greenlight: Well if jkroon can make it cope with .za internet connectivity then I am really quite optimistic about your prospects. |
| 12:21.05 | Greenlight | Did I mention we're using a 4G USB dongle... :) |
| 12:22.07 | Greenlight | Still for the most part latency seems 35-80ms, but rather jittery |
| 12:22.36 | Chainsaw | Greenlight: Probably HSDPA/UMTS handover. |
| 12:22.40 | *** join/#asterisk wonderworld (~ww@dslb-088-078-146-083.088.078.pools.vodafone-ip.de) |
| 12:23.10 | Chainsaw | Greenlight: Worth ensuring your traffic stream is continuous, as the cell site will otherwise step you down. |
| 12:23.11 | Greenlight | Isn't HSDPA 3G ? |
| 12:23.26 | Chainsaw | Greenlight: HSDPA is 3G, yes. |
| 12:23.56 | Chainsaw | Greenlight: If you get actual LTE, then it's still stepping down, probably to HSUPA or even further. |
| 12:24.48 | Greenlight | At that latency I assumed it was actual LTE |
| 12:24.48 | file | Black Friday becomes Cyber Monday. |
| 12:25.42 | Greenlight | Traffic stream is pretty constant around the 250k in and out levels just now |
| 12:25.57 | Greenlight | Occosional spikes in latency to 150ms though are worrying |
| 12:26.08 | Greenlight | Cyber Monday ? |
| 12:26.16 | file | Greenlight, shopping |
| 12:26.28 | Greenlight | Ah |
| 12:27.26 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:27.28 | Chainsaw | Greenlight: We're in some corporate deal with Vodafone, so 4G doesn't exist for me yet :/ |
| 12:28.11 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:28.13 | Greenlight | They've started their rollout. Me and the wife also both with Vodafone, and annoyingly she's started to get 4G signal, but I have not |
| 12:28.22 | Greenlight | Guess it's not enalbed on my account or something |
| 12:28.36 | Chainsaw | Greenlight: They knocked out Peterborough city centre for days due to what they said was 4G prep work. |
| 12:28.48 | Chainsaw | Greenlight: But still the best I've ever seen on my phone is an H+ indicator. |
| 12:29.00 | *** join/#asterisk jackbrown (~se@93-44-52-189.ip95.fastwebnet.it) |
| 12:29.02 | jackbrown | hello |
| 12:29.11 | Chainsaw | Good afternoon Jack. How can we help? |
| 12:29.40 | Greenlight | Apparently I even get coverage at my house, which considering i'm yeasr from the "rural" FTTP rollout, might be good news if they allow large usage caps soon |
| 12:30.46 | jackbrown | I'm trying to prioritize my VoiP Siemens telephone on my OpenWRT router firmewar and I'd would like to use asterisk |
| 12:30.54 | jackbrown | what do you think about ? |
| 12:33.17 | *** join/#asterisk Ibrahim22 (3ec2d0fa@gateway/web/cgi-irc/kiwiirc.com/ip.62.194.208.250) |
| 12:34.43 | Greenlight | You want to use Asterisk to priortise the traffic ? |
| 12:36.36 | Ibrahim22 | Hi, quick question on webrtc and asterisk 12.7.1, I have compiled and installed asterisk with ICE support (rtp debug shows Sent RTP packet to <my own computer ip>:51608 (via ICE) (type 00, seq 046151, ts 043360, len 000160)) I am using sipJS, but have tested both JsSip and sipml5 and no audio there either. How can i debug what is going on? |
| 12:39.14 | file | debugging WebRTC requires understanding it, and generally looking at the browser |
| 12:40.05 | file | did you follow the guide for setting it up? |
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| 12:44.59 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:49.48 | Ibrahim22 | Yes I did |
| 12:52.02 | file | can you provide a SIP log of a non-working call? |
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| 13:00.40 | Ibrahim22 | Alright! |
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| 13:03.01 | Greenlight | So if Im getting spikes of latency of say 150, compared to the normal of 50sih, bumping the jitterbuffer length up will help, yes? |
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| 13:17.09 | Ibrahim22 | https://gist.github.com/anonymous/62904ee3a09c86b4d89c |
| 13:19.34 | Ibrahim22 | That is the SIP log |
| 13:20.19 | Ibrahim22 | as you can see, Playback fails and the websocket is even closed before the BYE response is transmitted |
| 13:20.35 | file | looks fine from a signaling perspective |
| 13:20.41 | file | that's from the browser side |
| 13:21.27 | file | you'd have to look there to see what is causing it to do that |
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| 13:22.01 | Ibrahim22 | And what is 'that' and how and where do I look for it? |
| 13:22.28 | file | the web browser, the javascript there |
| 13:22.56 | file | it is what disconnected |
| 13:23.23 | Ibrahim22 | Does it help you if I told you sipJS gives the following errors: sip.invitecontext.mediahandler | ICE connection state changed to "failed" and sip.invitecontext.mediahandler | emitting event iceFailed |
| 13:24.10 | file | yes, it means the ICE negotiation in the browser probably failed - why, I don't know |
| 13:24.34 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 13:25.23 | marceloamorim | there is any way to remove acl like a module or something? |
| 13:25.39 | Ibrahim22 | So the ice settings in asterisk has nothing to do with it? |
| 13:26.00 | file | ICE is negotiated between both sides, it seems to have worked on the Asterisk side but not the browser side |
| 13:26.44 | Ibrahim22 | okay, I will turn on debugging in the browser and find out what is happening with the ICE in the browser |
| 13:27.48 | marceloamorim | I`m getting SIP Peer ACL: Rejecting '192.168.3.35' due to a failure to pass ACL '(BASELINE)' |
| 13:33.16 | marceloamorim | o.O I make this problem, I figured out |
| 13:33.17 | marceloamorim | sorry |
| 13:33.39 | Greenlight | Will enabling trunking on a IAX peer with active calls cause issues? Or will the trunking take effect for new calls only? |
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| 14:52.51 | v0lZy | Hello everyone |
| 14:53.11 | v0lZy | I have a situation in which I'm unsure as to what my sip.conf defined account should be set up like |
| 14:53.34 | v0lZy | I have an asterisk box behind a firewall, with several phones in the same subnet as the pbx |
| 14:54.08 | v0lZy | so its internet ->>> firewal->>>>[switch to which phones and asterisk is connected] |
| 14:54.26 | v0lZy | Now I've configured an OpenVPN server on the firewall itself |
| 14:54.40 | v0lZy | so incoming VPN connections create a tunnel between the roadwarrior and the firewall |
| 14:55.04 | v0lZy | at the roadwarrior end, i have a softphone |
| 14:55.27 | v0lZy | and I can register it to asterisk and i SIP goes through fine (phones ring) |
| 14:56.38 | v0lZy | but RTP doesnt seem to go through or rather, the roadwarrior with the softphone can hear the person they call, but the person that the roadwarrior calls can't hear the roadwarrior... |
| 14:57.04 | v0lZy | Now I'm unsure what my SIP settings should be in this scenario... what I suspect is that nat=yes might have something to do with this here |
| 14:57.47 | v0lZy | openVPN tunnel endpoint is 10.255.255.5 ... its communicating to 10.255.255.6 (the firewall) which then routes to 192.168.1.3 (the asterisk box) |
| 14:58.32 | tomodachi | v0lZy: well if you are using vpn and routing then nat should be no |
| 14:58.43 | v0lZy | On the firewall, I'm seeing UDP traffic from the roadwarrior to the public WAN IP (so not within the VPN tunnel) which leads me to beleive that my asterisk gives the wrong IP address to the roadwarriro to which to open RDP ports to. |
| 14:58.44 | tomodachi | preferably your vpn should use routing instead of nat |
| 14:59.03 | v0lZy | tomodachi-san! |
| 14:59.25 | v0lZy | I have no idea of how OpenVPN handles this internally |
| 14:59.32 | v0lZy | I imagine it routes |
| 14:59.34 | tomodachi | by default its routing |
| 14:59.49 | v0lZy | ok, so if i set nat=no for that sip account |
| 14:59.52 | tomodachi | unless you use bridging , your firewall of course could nat if configured improperly |
| 15:00.27 | v0lZy | might be that its doing NAT |
| 15:00.35 | tomodachi | you can easily check |
| 15:00.48 | tomodachi | se if you can access the clients vpn ip from the pbx |
| 15:01.17 | tomodachi | like testing accessing the clients vpn ip over ssh |
| 15:02.27 | v0lZy | it cant. |
| 15:02.55 | v0lZy | ip is 10.255.255.6, can't ping it from 192.268.1.3 |
| 15:04.37 | v0lZy | i suppose I can add a NAT rule which has a destination ... |
| 15:04.48 | v0lZy | to NAT to that IP rather than WAN ip.. |
| 15:06.05 | v0lZy | tomodachi: does in that case 'nat=yes' apply though? |
| 15:06.11 | v0lZy | thing is if my firewall is doing outbound NAT |
| 15:06.21 | v0lZy | so i can talk to my provider and accept calls etc |
| 15:06.28 | v0lZy | they this is kind of similalr, no? |
| 15:07.26 | v0lZy | problem being that if im doing outbound nat for certain ports coming from my asterisk and going to my provider... then thats probalbly no different for a softphone on the other end... asterisk i asume doesnt care if its provider's sip/rtp coming in or some phone's? |
| 15:09.27 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:13.49 | tomodachi | voxter: you dont really want nat between clients and pbx since SIP and nat traversal is a mess |
| 15:14.08 | tomodachi | so i would disable nat in sip config, make sure my firewall does not NAT to vpn subnets |
| 15:14.14 | tomodachi | only to wan |
| 15:15.00 | v0lZy | tomodachi: let me draw a little diagram... one moment please |
| 15:16.21 | WIMPy | v0lZy: Probably one side is trying to be clever and try to use the direct way without the VPN. Just the normal issues. |
| 15:16.48 | tomodachi | is your pbx on an internal ip or public ip?! |
| 15:16.50 | tomodachi | v0lZy: |
| 15:21.56 | v0lZy | Hi WIMPy |
| 15:22.01 | v0lZy | http://snag.gy/Zlw8J.jpg |
| 15:22.23 | v0lZy | WIMPy, yeah i think thats what my VPN roadwarrior is doing, but its probably doing that cause asterisk tells it thats the IP it should send rtp to, no? |
| 15:22.30 | v0lZy | tomodachi: my PBX is on an internal IP |
| 15:22.32 | v0lZy | check the diagram |
| 15:23.19 | WIMPy | Likely. SIP debug will tell you more. |
| 15:23.33 | WIMPy | Did you define the VPN as local net? |
| 15:24.13 | v0lZy | I havent told asterisk anything about the VPN |
| 15:24.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:24.29 | v0lZy | asterisk's default route is 192.168.1.1, which is the firewall that acts as a NAT and gateway |
| 15:24.52 | v0lZy | 10.255.255.0/24 being the roadwarrior subnet, asterisk just sends it upstream to the firewall and the firewall's supposed to be handling the routing to that |
| 15:25.07 | v0lZy | however, if the RTP ports negotiation is sent with the public IP |
| 15:25.21 | WIMPy | But you need to tell Asterisk that it's not behind NAT. |
| 15:25.22 | v0lZy | then vpn tunnel wont be used by the softphone |
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| 15:25.40 | v0lZy | i didnt know there was a way to tell asterisk which networks to nat and which not too |
| 15:25.44 | v0lZy | or maybe i knew that and forgot... :D |
| 15:25.47 | WIMPy | Or just switch to an IAX phone :-) |
| 15:26.09 | WIMPy | It needs to know. |
| 15:27.15 | tomodachi | voxter: by default i presume asterisk will assume its NOT nat |
| 15:27.26 | WIMPy | OTOH, the VPN doesn't route to your Asterisk, if I get your picture right. So that would be NAT? |
| 15:27.32 | tomodachi | voxter: still can you connect to some service on vpn endpoint from asterisk? (to verify that its not nat:ed) |
| 15:28.04 | WIMPy | How does that client get to Asterisk? |
| 15:28.20 | v0lZy | the vpn terminates at the firewall |
| 15:28.47 | v0lZy | firewall internall then routes the traffic to the targeted 192.168.1.3 ip (thats the ip i use in the softphoen to reach asterisk) |
| 15:28.55 | WIMPy | And how does it take the last hop? |
| 15:28.57 | v0lZy | and indeed the soft phone computer cna ping asterisk |
| 15:29.07 | WIMPy | So it's port forwarding? |
| 15:29.24 | v0lZy | id guess its routing |
| 15:29.32 | v0lZy | port forwarding is public IP to asterisk |
| 15:29.42 | v0lZy | my ISP brings the sip trunk to me |
| 15:29.50 | v0lZy | and i port map it to asterisk |
| 15:30.11 | WIMPy | So what IP does the phone talk to? |
| 15:30.21 | v0lZy | tomodachi: hm, not running any services on the roadwarrior side. |
| 15:30.29 | v0lZy | phone talks to 192.168.1.3 |
| 15:30.37 | tomodachi | v0lZy: can the roadwarrior ssh to asterisk? |
| 15:30.41 | v0lZy | which is the LAN IP of asterisk, which only has one IP (that one) |
| 15:30.50 | WIMPy | Ok, yes, that looks routed. |
| 15:30.50 | v0lZy | tomodachi let me try that |
| 15:31.22 | v0lZy | tomodachi: affirmative, I can ssh into asterisk as roadwarrior |
| 15:31.28 | WIMPy | So the VPN is a local net then. |
| 15:31.30 | tomodachi | so its routed then ,its good |
| 15:31.37 | tomodachi | should work :) |
| 15:31.44 | v0lZy | like i said, I got sip runing, this is just an RTP thing and what i think is going on is that asteriks is negotiating rtop ports but advertising public ip |
| 15:31.48 | v0lZy | rtp* |
| 15:32.15 | v0lZy | WIMPy: from the perspective of asterisk, the VPN subnet is a network it reaches through the 192.168.1.1 gateway |
| 15:32.24 | v0lZy | they are two separate subnets |
| 15:32.43 | v0lZy | asterisk relies on the firewall (192.168.1.1) form its perspective, ro locate 10.255.255.6 (the vpn roadwarrior) |
| 15:32.57 | WIMPy | Check your localnet settings |
| 15:33.07 | v0lZy | firewall has many ips... 192.168.1.1 is the one exposed to the asterisk, 10.255.255.5 is the one exposed to the vpn roadwarrior |
| 15:33.21 | v0lZy | WIMPy: what file is that parameter in? I can't find it |
| 15:33.40 | WIMPy | sip.conf |
| 15:33.49 | v0lZy | doh, enver mind just found it |
| 15:33.53 | v0lZy | can i comma separate multiple subnets? |
| 15:34.01 | v0lZy | right now my localnet=192.168.1.0/24 |
| 15:34.09 | WIMPy | Err, check the sample. |
| 15:34.31 | WIMPy | IIRC just use it multiple times. |
| 15:34.52 | v0lZy | i see |
| 15:34.54 | v0lZy | just one after another |
| 15:34.55 | v0lZy | so |
| 15:35.21 | v0lZy | if i add 10.255.255.0/24, Asterisk will not rewrite its public IP into the packets when sending rtp port negotiations in sip? |
| 15:35.27 | v0lZy | sounds like what i need |
| 15:36.23 | WIMPy | That's the idea anyway. |
| 15:36.48 | v0lZy | yeap works |
| 15:36.50 | v0lZy | mistery solved |
| 15:36.53 | v0lZy | you guys are the best |
| 15:36.58 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 15:37.00 | tomodachi | woho, i also learnt something along the way! |
| 15:37.08 | v0lZy | tomodachi: makoto ni kokoro kara arigatou gozaimashita |
| 15:37.21 | tomodachi | v0lZy: dozo! |
| 15:38.12 | v0lZy | Its rare to come accross japanese speakers :D |
| 15:38.27 | v0lZy | WIMPy: this localnet parametr is darn useful |
| 15:38.40 | tomodachi | i would hardly call my self a japanese speaker though , but ive been there a few times so some things caught on |
| 15:38.44 | v0lZy | I assume my sip.conf should state nat=no in this case... |
| 15:38.52 | WIMPy | More like neccessary to get SIP working at all. |
| 15:39.01 | v0lZy | though thwats the harm in leaving it nat=yes if localnet parameter defines the ip tis coming from as localnet |
| 15:39.12 | v0lZy | i suppose asterisk ignores nat=yes if the network the conenction is coming in from is localnet |
| 15:39.52 | WIMPy | Not exactely sure what it does. It still says Transmitting (NAT), but it doesn't seem to do any harm. |
| 15:40.43 | WIMPy | Since Asterisk told me that there might be a security issue in not having all peers the same, I removed all nat=no's and that's working fine. |
| 15:42.33 | v0lZy | think the world will move to IAX instead of SIP any time soon? this RTP thing is kind of annyoing along the lines of FTP |
| 15:43.09 | WIMPy | Unfortunatley the ones who invented it doen't seem to think much of it :-( |
| 15:43.26 | WIMPy | But it's obviousely a lot better in many ways. |
| 15:43.52 | WIMPy | Well, anything publicly documented is better than SIP. |
| 15:46.05 | v0lZy | i wonder why when they were designing SIP they didnt just kind of negotiate VLAN like tags on packets and instead went for RTP prots |
| 15:46.06 | v0lZy | ports* |
| 15:46.43 | WIMPy | How would that work? |
| 15:47.19 | v0lZy | well the way I understand the situation is that RTP ports are used so that you get simultanious conversations |
| 15:47.39 | v0lZy | you and me talk on a set of ports and tomodachi and my coworker talk on a different set of rtp ports |
| 15:47.43 | WIMPy | They carry the voice data, yes. |
| 15:48.17 | v0lZy | they could carry the voice data over the same port... a web server only uses port 80 |
| 15:48.23 | WIMPy | They wouldn't have to be different. |
| 15:48.50 | v0lZy | dont understnad why voip couldnt be done that way |
| 15:49.13 | v0lZy | instead of using ports, you'd just negotiate a tag ... like a session-id |
| 15:49.20 | v0lZy | and that would be it... |
| 15:49.26 | WIMPy | It can be done. There are better solutions available. You just mentioned one yourself. |
| 15:49.35 | WIMPy | RTP does have them. |
| 15:50.38 | v0lZy | wonder what the reason was for not making it that way and choisng sip/rtp combo... |
| 15:51.14 | WIMPy | I don't have the impression that SIP was really done for VOIP. |
| 15:53.02 | WIMPy | And it still hasn;t really been shoehorned in, yet. |
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| 15:59.03 | v0lZy | I shouldn't really cry about it, gives us IT people stuff to do that others shy away from ;) |
| 15:59.52 | WIMPy | I'd prefer not to have to do it. I like things that simply work. |
| 16:02.28 | v0lZy | and hold it as a secret? :D |
| 16:02.35 | v0lZy | best have everything doen automagically |
| 16:02.50 | v0lZy | and pretend its super complex to set up and configure ;D |
| 16:03.20 | v0lZy | "This thing's so difficult I'm on my second beer break already and I still can't get it to work the way we want!" |
| 16:03.22 | v0lZy | </bluff> |
| 16:03.33 | WIMPy | It IS very comlex. |
| 16:04.24 | WIMPy | Many months of wasted time. |
| 16:04.33 | v0lZy | im speaking hypothethically if it were automagic... :D |
| 16:04.55 | WIMPy | An who does the automation? |
| 16:05.19 | v0lZy | while a super simple setup wizzard of course! |
| 16:05.27 | v0lZy | next next next close |
| 16:05.28 | v0lZy | :D |
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| 16:11.16 | WIMPy | That's the reason why I'm sure a classic PBX is much cheaper in the end than using Asterisk. |
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| 16:30.01 | v0lZy | if you count in human costs, then yeah, probably |
| 16:30.17 | v0lZy | but then again, the hardware is expensive for classic pbx |
| 16:30.29 | v0lZy | then again, if you want a drop in replacement, ippbx isnt much cheaper either |
| 16:30.35 | v0lZy | poe switches, UPSes |
| 16:30.38 | WIMPy | Not more than the new stuff. |
| 16:30.57 | WIMPy | Or TDM cards. |
| 16:31.02 | v0lZy | id adds good to existing infrastructure, ubt if yyou have to build it from scratch... high quality links, firewalls, servers.. probably amounts to more rather than less |
| 16:31.22 | v0lZy | then again |
| 16:31.28 | v0lZy | everyone has a mobile phone |
| 16:31.34 | v0lZy | wifi + asterisk + softphone.. |
| 16:31.50 | v0lZy | if you already have a qulified admin |
| 16:31.53 | v0lZy | on a flatrate |
| 16:31.57 | WIMPy | And the shit draws quite excessive amounts of power, so it stays expensive. |
| 16:32.19 | WIMPy | Wifi for voive is not a good idea. |
| 16:32.38 | WIMPy | Unless you don't use it for anything else and noone else in the neighbourhood does either. |
| 16:33.03 | v0lZy | just a dedicated vlan on wifi |
| 16:33.09 | v0lZy | all depends on what kind of stuff you already have |
| 16:33.16 | v0lZy | i work with several companies of different ages |
| 16:33.26 | WIMPy | How is that supposed tohelp? |
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| 16:33.31 | v0lZy | 1 is big and new and has a big mother company behind it |
| 16:33.37 | v0lZy | the other is small and new |
| 16:33.43 | v0lZy | then theres two established.. one medium one big |
| 16:33.54 | v0lZy | quite different takes on IT expenses |
| 16:34.15 | v0lZy | those that plan their budget ahead to say 10 000 eur ... to them its a yearly project |
| 16:34.24 | v0lZy | those that plan below that ... they just tend to outsource and not worry about it |
| 16:34.48 | v0lZy | and those that plan higher than that, lets say in the 50 000 eur range, to them its a side projects |
| 16:34.51 | v0lZy | project* |
| 16:34.58 | v0lZy | they usually already have good switches, good links etc |
| 16:35.17 | v0lZy | where as small companies with barely a server struggle on deciding what ups to buy :D |
| 16:36.15 | v0lZy | the difference i want to point out is the synergy of what u use all this stuff for |
| 16:36.30 | v0lZy | if you have it only to support voip, then its definetly more expensive |
| 16:37.03 | v0lZy | but if you have lots of this infrastrucutre bcause of other reasons .. .like u already have good switches, a good firewall, good links, wifi in all the offices etc.. |
| 16:37.09 | v0lZy | maybe even a couple of different locations |
| 16:37.23 | v0lZy | then its probably a lot cheaper |
| 16:37.31 | WIMPy | Yes, but you can only use it for multiple things if you have good switches and the man power to configure them propperly. |
| 16:37.55 | WIMPy | Otherwise you're better off setting up independant networks. |
| 16:39.12 | v0lZy | yeah, off the shelf benefits there too if you go that way |
| 16:39.27 | v0lZy | but then it doesnt scale well horizontally |
| 16:45.41 | hrnt | i don't know if this is the correct channel, but anybody have experience with polycom IP and VVX series phones? |
| 16:45.54 | hrnt | (or can let me know if there is a more suitable channel for this question) |
| 16:47.40 | WIMPy | Well, we don't know the question yet. But don't they have support channels? Anyway, there should be several Polycom fans around here. |
| 16:48.22 | hrnt | nah, not really looking for support, just experiences :) we have right now mostly IP 550's but we are planning on getting some new phones |
| 16:49.08 | hrnt | and the way i see it, we can stick with IP 550's or we could get new phones from the VVX range. I am mostly curious about how those phones are compared to e.g. IP 550. Is the speakerphone still as good as in the IP series? basically, is there any reason _not_ to upgrade? |
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| 16:50.01 | hrnt | and also is the touch screen any good or would it be a good idea to stick with VVX 410 with hard keys |
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