| 00:08.31 | volga629 | is possible set one var in dialplan with multiply values |
| 00:09.07 | [TK]D-Fender | as in? |
| 00:09.25 | volga629 | exten => _X.,n,Set(PEER_GRP=${DB(AMPUSER/${PEER_D}/device)}) |
| 00:09.28 | volga629 | exten => _X.,n(found),Set(PEER_GRP=${DB(AMPUSER/${EXTEN}/device)}) |
| 00:09.42 | volga629 | like if statement |
| 00:10.23 | volga629 | if something set var and else set same but different value |
| 00:11.09 | volga629 | here it processing both |
| 00:11.11 | volga629 | <PROTECTED> |
| 00:11.13 | volga629 | <PROTECTED> |
| 00:13.03 | [TK]D-Fender | You're overwriting obviously... |
| 00:13.07 | [TK]D-Fender | what would you do that? |
| 00:13.20 | [TK]D-Fender | Why not get the other DB value into ANOTHER variable? |
| 00:13.30 | [TK]D-Fender | Do you ahve any idea what you're doing? |
| 00:14.25 | volga629 | http://fpaste.org/79100/41640139/ |
| 00:14.46 | volga629 | I am testing if it is mobile extension or local |
| 00:15.39 | volga629 | mobile extension is 5 digits and local 3 based on this I am setting var for while loop |
| 00:16.29 | [TK]D-Fender | volga629: paste the ENTIRE extension |
| 00:16.38 | [TK]D-Fender | not some broken middle piece only |
| 00:16.42 | volga629 | ok |
| 00:17.00 | [TK]D-Fender | no need really |
| 00:17.19 | volga629 | http://fpaste.org/79101/92941824/ |
| 00:17.20 | [TK]D-Fender | Because I can already see you aren't paying attention |
| 00:17.42 | [TK]D-Fender | exten => _X.,n,GotoIf($["${LEN(${PEER_F})}" = "5"]?look:found) <- here you just if it's found and do your "set" |
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| 00:18.02 | [TK]D-Fender | exten => _X.,n(found),Set(PEER_GRP=${DB(AMPUSER/${EXTEN}/device)}) <- this is where it goes if found |
| 00:18.03 | [TK]D-Fender | BUT... |
| 00:18.15 | [TK]D-Fender | if it is NOT found.... it calls exten => _X.,n,Set(PEER_GRP=${DB(AMPUSER/${PEER_D}/device)}) |
| 00:18.23 | [TK]D-Fender | and tehn CONTINUES to the next set |
| 00:18.26 | [TK]D-Fender | YOU didn't stop it |
| 00:19.06 | volga629 | or I see that make sense so instead found set the var and leave look |
| 00:19.23 | [TK]D-Fender | you let one roll right through |
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| 00:41.40 | volga629 | thank you for help it working right now |
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| 00:47.26 | volga629 | Do you know if MessageSend should accept sip:101&10101 |
| 00:47.29 | volga629 | ? |
| 00:47.47 | [TK]D-Fender | of course not |
| 00:47.51 | [TK]D-Fender | there is no "multiple" |
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| 01:01.02 | volga629 | xmpp account changing state " On the phone " as expected, but SIP SIMPLE how need set state if user on the phone ? |
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| 01:02.41 | [TK]D-Fender | huh? |
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| 01:08.27 | volga629 | I am asking about device state through SIP SIMPLE |
| 01:09.23 | [TK]D-Fender | * doesn't poll SIMPLE |
| 01:09.46 | [TK]D-Fender | If you want to know if a device is used on a call, * can only monitor calls & peers it manages |
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| 01:09.55 | [TK]D-Fender | And that has nothing to do with SIMPLE |
| 01:12.44 | volga629 | I see |
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| 01:33.43 | SupaYoshi | Hi i was looking for some advice on using a SIP client, (my android device) as a SIP client outside of my LAN. |
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| 01:34.25 | SupaYoshi | I have a strong password / secret for the sip client. However I have a NAT, and have port 5060 open for SIP traffic to my PBX. |
| 01:34.40 | SupaYoshi | However, I cant register to asterisk server, and even so. I was thinking, I also have a VPN server |
| 01:34.55 | SupaYoshi | would it be advised to only let SIP clients connect through VPN or the local LAN? |
| 01:34.58 | SupaYoshi | To improve security? |
| 01:35.22 | SupaYoshi | Or is it okay to connect phones over the internet? I mean I don't believe my calls would need to be encrypted. |
| 01:35.34 | SupaYoshi | But having registration attempts from the internet doesn't sound so secure? |
| 01:35.58 | SupaYoshi | I don't know please shoot me with some tips / do's don'ts, :D |
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| 01:58.59 | newtonr | SupaYoshi, http://svnview.digium.com/svn/asterisk/branches/11/README-SERIOUSLY.bestpractices.txt?view=markup |
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| 02:03.58 | newtonr | SupaYoshi, If you can setup your SIP clients over VPN and LAN only, then that is preferable |
| 02:04.18 | newtonr | SupaYoshi, other than that doc, look at the permit,deny,acl options in sip.conf |
| 02:04.59 | newtonr | see the tips right at the top of sip.conf sample file as well |
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| 02:31.55 | SupaYoshi | tyuh |
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| 05:37.47 | Grogdor | hi, this is not really asterisk-related but im wondering if you can enlighten me... |
| 05:38.07 | Grogdor | ive got an ht702 adapter doohickey logged into two extensions with callcentric |
| 05:38.32 | Grogdor | and im seeing a constant 7Kbps of traffic from it, is that normal? what's it doing? |
| 05:39.02 | Grogdor | the adapter has like 40,000 advanced options that i dont understand, just followed the callcentric config guide |
| 05:41.04 | [TK]D-Fender | Should use that BW without a channel being up.\ |
| 05:41.10 | [TK]D-Fender | Check your firewall |
| 05:44.23 | Grogdor | should or shouldnt? |
| 05:45.40 | [TK]D-Fender | sholdn't* |
| 05:45.47 | [TK]D-Fender | can't type tonight.. |
| 05:48.42 | Grogdor | http://pastebin.com/iTzNYX32 |
| 05:48.45 | Grogdor | that. constantly. |
| 05:50.29 | [TK]D-Fender | dump that actual packets |
| 05:53.51 | Grogdor | that's all it is, looks like a binding request/response for STUN to get thru my NAT |
| 05:54.11 | Grogdor | which is... slightly excessive... but gives me some idea of wher eto start fixing it |
| 05:55.44 | [TK]D-Fender | Should the revelation that the problem is stun.. be a surprise? </pun> |
| 05:56.17 | Grogdor | haha. not one bit, already suspected it ;) since i hadnt setup the nat redirect on this new firewall |
| 05:57.30 | [TK]D-Fender | you generally don't need one. |
| 05:58.00 | [TK]D-Fender | even a modest nat keep-alive is practically nothing... like a packet every 30 seconds or so just to keep it going |
| 05:58.30 | Grogdor | hm, you're right, it's commented out in my old rules |
| 05:58.33 | Grogdor | so why the hell |
| 05:58.45 | Grogdor | oh right.. was trying to fix something that i broke.. |
| 05:59.58 | Grogdor | stun is off, stun server isnt set, keepalive is 20sec.. dafuq |
| 06:00.36 | Grogdor | oh, it's set per port as 'keep-alive' |
| 06:02.00 | Grogdor | well this'll be fun. thanks! |
| 06:02.02 | Grogdor | gnite |
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| 10:53.19 | As001 | hello, what is the proper way to diagnose real packet loss in sip+rtp communication with mtr ? I am using it like this mtr -i 0.02 -s 200 -c 1000 ipaddress, Is interval of 0.02 and size of 200 ok to sumulate voip traffic which use alaw codec ? |
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| 13:15.47 | justa | Hey there. |
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| 13:27.03 | anonymouz666 | still searching for a way to debug audio problems with chan_dahdi |
| 13:30.24 | justa | I've got a PIAF/Incredible PBX running on a PI; worked fine from YATE.. But i have issues with CSipSimple. 488-not-accepted-here issues whatever I try to call any of the 'feature codes' like voicemail, echo, time-of-day, etc. Checking to see what might be wrong, I see that in 'sip show settings', there's gsm,pcmu and speex available. In CsipSimple those are now the only codecs i've configured; all at 8Khz (though i've had them as high as 16 and 32khz ... |
| 13:30.30 | justa | ... too). No matter what I do, the asterisk-server reports a "SIP/2.0 488 Not acceptable here" and destroys the sip-session. |
| 13:31.19 | justa | i have set 'sip set debug on' to watch what goes on concerning SIP and Asterisk<-->CSipSimple; but is there a way to see *why* asterisk decides there's no compatibility of codecs possible ? |
| 13:31.27 | justa | (another debug channel i can set ?) |
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| 13:46.51 | justa | weird shit.. |
| 13:47.00 | justa | Just now tried with 'sipdroid' and it works on the first try |
| 13:48.36 | kaldemar | justa: sip debug and configs for the matching device in sip.conf |
| 13:48.54 | kaldemar | or some other place where piaf stuffs the configs. |
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| 13:53.42 | justa | kaldemar: thanks. I think i'll check the SipDroid codec-list and see if I can get CSipSimple to match. |
| 13:53.56 | P-NuT | Hi all, Can anyone tell me how to verifiy the users connecting to asterisk via TLS? |
| 13:54.28 | P-NuT | I have investigated the tlsverifyclient=yes method in asterisk 1.8, but it doesn't appear to make a blind bit of difference |
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| 14:02.31 | justa | kaldemar: the main difference I seem to experiencing is that in 'sip debug', with 'sipdroid' i get a 'capabilities' line to show which codecs match between both parties. With CSipSimple, this never seems to happen; not even a 'no match' or anything |
| 14:04.21 | kaldemar | there should be a difference in the invites. |
| 14:06.48 | justa | ack. It's what I'm concentrating on now. |
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| 14:22.41 | tulga | what is different conf-adminmenu and conf-adminmenu-162 and conf-adminmenu-18 files? |
| 14:27.57 | [TK]D-Fender | tulga: confbridge changed a lot from 1.6 to 1.8, and again a lot goign to 10, then again for 11 |
| 14:28.51 | tulga | [TK]D-Fender: now I have 2.11.0.3 version. what audio file used? |
| 14:29.04 | [TK]D-Fender | that is not an Asterisk version |
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| 14:29.25 | tulga | [TK]D-Fender: yes, it is conference module version. asterisk 11.7 |
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| 14:29.50 | [TK]D-Fender | that is FREEPBX which means nothing here |
| 14:30.07 | [TK]D-Fender | Asterisk 11.7 is not 1.6 and it is not 1.8 |
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| 14:48.08 | Kobaz | i have a fun question |
| 14:48.40 | Kobaz | not really specificly an asterisk one, but i wonder if there's a way to get a core file from asterisk, without actually killing asterisk, and without interupting it either |
| 14:48.45 | Kobaz | so no kill -sigquit |
| 14:48.49 | Kobaz | and no gdb |
| 14:49.06 | Kobaz | well, no gdb process attach leaving it paused |
| 14:51.29 | Kobaz | oh cool, found it |
| 14:51.34 | Kobaz | http://stackoverflow.com/questions/68160/is-it-possible-to-get-a-core-dump-of-a-running-process-and-its-symbol-table |
| 14:51.37 | Kobaz | yay google |
| 14:54.18 | blitzrage | Kobaz: oh nice |
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| 14:57.04 | Kobaz | you still attach, but it's quick |
| 14:57.14 | Kobaz | so it should be minimal pausing of the running app |
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| 15:24.32 | anonymouz666 | sruffell: is there a way to look for incoming audio in DAHDI channels? I've got some "mute calls" from telco, I want to "prove" that telco doesn't send any audio at all. at signalling level I don't have any problem |
| 15:25.07 | anonymouz666 | I was looking for dahdi_pcap but it was designed for SS7 links? |
| 15:26.48 | BeachBall | anyone use twilio ? |
| 15:29.23 | sruffell | yeah, dahdi-pcap is for hdlc / d-channels. You will need to use dahdi_monitor in order to record the audio from the cards. |
| 15:29.52 | sruffell | You will need to figure out which b-channel is used for the call, and then run dahdi_monitor on that channel. |
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| 15:32.13 | anonymouz666 | I need to open 30 SSH sessions and run 30 times dahdi_monitor then :P |
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| 15:33.31 | anonymouz666 | OK, a shell script can help me on that :-) |
| 15:34.14 | sruffell | heh..yeah…one ssh session and a "for chan in {1..15} {17..30}; do dahdi_monitor $chan -r /tmp/chan-${chan}.raw &; done |
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| 15:35.31 | sruffell | or something like that…I see I put an extra ';' in there. |
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| 16:48.29 | workingcats | sruffell, looks like the right amount of ; to me, where's the spare one? |
| 16:48.41 | sruffell | after the '&' |
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| 17:02.40 | justdave | sure is confusing trying to update dahdi on a rhel5 system using digium's official repos |
| 17:02.56 | justdave | newest version of kmod-dahdi-linux is for an older kernel |
| 17:03.07 | justdave | and the second-newest kmod-dahdi-linux is for the current kernel |
| 17:03.28 | justdave | so yum tries and fails to downgrade your kernel if you just let it grab the newest version |
| 17:03.58 | justdave | had to manually install the second-newest one that matched the new kernel and then exclude it from the yum update |
| 17:04.53 | justdave | likes the rhel6 packages better, that use dkms, so you don't have to worry about matching a kernel anymore |
| 17:05.20 | justdave | and hopefully soon I'll be rid of these rhel5 boxes |
| 17:06.05 | justdave | on of them goes away in 3 weeks, the other goes away whenever someone figured out how to ship a new box to China and find someone there that knows how to set it up :) |
| 17:07.16 | navaismo | hint: use the sources |
| 17:07.27 | justdave | not allowed to |
| 17:07.29 | justdave | opsec won't let me |
| 17:07.42 | navaismo | too bad |
| 17:07.56 | justdave | well, I could build my own RPMs, but it has to be rpm to go on the box |
| 17:09.27 | justdave | mainly in order to not have any build tools on the production boxes |
| 17:12.28 | justdave | I used to do all the asterisk stuff from source before we had that rule, many years ago :) |
| 17:25.56 | Kobaz | -rw------- 1 pbx pbx 3.0G Jun 13 2012 6139.1339630372.asterisk.core |
| 17:25.57 | Kobaz | haha |
| 17:25.58 | Kobaz | yay |
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| 17:51.02 | *** join/#asterisk dingle (dingle@secretpolice.net) |
| 17:51.31 | dingle | hello, i am just starting out using asterisk. what is a good provider to use? |
| 17:52.27 | workingcats | dingle, you mean to make calls into the normal phone networks? |
| 17:52.33 | dingle | yes |
| 17:52.41 | workingcats | might help to tell us where you are then ;) |
| 17:52.45 | dingle | the one i was looking at begins with a V |
| 17:52.48 | dingle | i am in USA |
| 17:52.57 | justdave | home or business? |
| 17:53.01 | dingle | it costs $7.95 a month |
| 17:53.07 | dingle | home for now, then business |
| 17:53.59 | justdave | I use Broadvoice for my home connection. We use Level 3, Wiline, and OnSIP at work. |
| 17:54.20 | justdave | Level 3 and Wiline are enterprise-level stuff, OnSIP has good small-business stuff (and home stuff too) |
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| 17:54.47 | justdave | Level 3 also sucks if you want anything done in a timely manner. |
| 17:55.09 | justdave | they're one of those "ok, let's take 2 weeks to do this paperwork, and then we'll have your login creds for the SIP trunk ready in a month or so" places. |
| 17:55.29 | workingcats | justdave, lol :D |
| 17:55.38 | dingle | onsip seems like too much |
| 17:55.55 | dingle | man, i wish i could remember the name of this provider |
| 17:56.32 | justdave | not Vonage is it? |
| 17:56.36 | [TK]D-Fender | Voicepulse |
| 17:56.41 | justdave | (don't use Vonage :) |
| 17:58.25 | dingle | a friend of mine used vonage, it was total junk |
| 17:58.31 | mmikeym | can 911 typically tell if your caller ID is spoofed? |
| 17:58.42 | dingle | its not voiceplus |
| 17:59.24 | justdave | we had a problem with call capacity during conference calls on our Level 3 line (it was limited to 24 concurrent calls). It took me 2.5 months to get them to set up a failover so it would deliver that number to one of our other offices when it ran out of slots in that one. |
| 17:59.56 | dingle | it had the ability to send sms |
| 18:00.33 | dingle | and of course i cant find it in my history |
| 18:00.38 | justdave | mmikeym: not quite sure how it works in the VoIP world, but I know on landlines 911 doesn't use CallerID at all, they get the E911 data related to the subscriber from the phone company separately |
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| 18:00.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:01.21 | justdave | I think for most VoIP providers it's the same, which is why most of them have you configure the E911 separately after you sign up |
| 18:01.24 | mmikeym | justdave: yeah right now with my voip I just have the caller ID set to my physical landline which I thought would be enough for 911 service but I'm not so sure...I suppose I should setup E911 with VoIP just in case |
| 18:01.38 | dingle | found it, its vitelity |
| 18:02.01 | justdave | mmikeym: if they don't say they offer E911, 911 might not even work. |
| 18:02.16 | justdave | I'm pretty sure if they give you a real phone number they're required to provide it now though. |
| 18:02.48 | mmikeym | justdave my voip provider does offer E911 but I never set it up...we still have a physical line coming into the builiding from a POTS provider so I figured using that caller ID would provide the address if we made an outbound 911 call |
| 18:02.58 | justdave | usually you have to fill out extra stuff on your billing portal to set it up after you subscribe though (to verify your physical location that'll be reported when you call) |
| 18:03.06 | dingle | i got a fairly nice phone to use, its a cisco 7965 |
| 18:03.13 | justdave | also, most VoIP providers nowadays won't let you fake your callerID anymore |
| 18:03.42 | justdave | except for the ones that like to make money off the scammers that don't want to get caught :) |
| 18:03.47 | mmikeym | we set the caller ID through asterisk and it seems to pass through just fine |
| 18:03.58 | justdave | like "Rachel from Card Services" |
| 18:04.02 | dingle | does anyone use vitelity? |
| 18:04.38 | workingcats | justdave, depends on type of contract, too |
| 18:05.17 | workingcats | or it can at least |
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| 18:05.31 | justdave | yeah, I've seen it allowed for businesses so they can use the "find me/follow me" stuff to forward calls to a cell phone and still have the original caller's callerID on it. |
| 18:05.55 | workingcats | a normal home "line" should have the ID enforced by the provider, but for certain business types at least it makes sense to permit the customer to set the id |
| 18:06.32 | [TK]D-Fender | dingle: Vitelity? |
| 18:06.41 | [TK]D-Fender | nm, just scrolled up |
| 18:07.06 | dingle | yes sir |
| 18:07.25 | dingle | is it any good? or are there cheaper/better ones out there? |
| 18:07.25 | mmikeym | most wholesale voip providers still allow you to set the caller ID though I thought? |
| 18:08.10 | workingcats | mmikeym, i think i can set anything on both of ours |
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| 18:09.41 | dingle | are there any providers that allow unlimited usa continental in/out bound calls? |
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| 18:15.01 | [TK]D-Fender | many |
| 18:17.02 | mmikeym | the usual stipulation is that its for home use only though...the unlimited lines I've seen |
| 18:17.10 | dingle | would you please direct me to a couple? |
| 18:17.41 | [TK]D-Fender | How much do you really expect to be calling? |
| 18:18.01 | dingle | 1000 minutes a month |
| 18:18.18 | mmikeym | I use voip.ms...been fairly happy with them..I think when you buy a DID they have an unlimited option but I never use it since per minute is so cheap anyway |
| 18:18.38 | [TK]D-Fender | might not be worth it...I get rates easily towards $0.01/min .. that's $10..... |
| 18:18.54 | dingle | where at? |
| 18:18.59 | [TK]D-Fender | Shop around |
| 18:19.06 | [TK]D-Fender | ~itsplist-us |
| 18:19.06 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:20.51 | dingle | i am trying to shop around, i am just asking for services people use o0n a regular basis |
| 18:21.11 | dingle | so i dont end up with something i hate |
| 18:24.37 | [TK]D-Fender | come back with 2-3 and we'll vet them |
| 18:25.14 | BeachBall | how much member does my box need per call? |
| 18:25.18 | BeachBall | memory* |
| 18:25.36 | BeachBall | haha i started to think, but i stopped |
| 18:26.32 | navaismo | depends what you do, there is no rule you must check againt your current arch by using top, free and others tools |
| 18:29.06 | dingle | i maxed mine out to 2gb ram |
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| 18:42.10 | dingle | i need a floppy drive and disk |
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| 18:52.57 | dingle | i believe teliax or voicepluse will suit my needs |
| 18:53.17 | dingle | anyone use either of these services? |
| 19:01.35 | dingle | hrm. |
| 19:07.25 | BeachBall | how can i test my pbx box for it's max connections? |
| 19:07.51 | BeachBall | who wants to call BeachBall ? |
| 19:07.52 | BeachBall | ;D |
| 19:07.53 | Penguin | for it is max connections? |
| 19:08.46 | dingle | BeachBall: i will help |
| 19:09.26 | [TK]D-Fender | fires up Qwell's chan_skinny botnet and prepares to target BeachBall's server |
| 19:11.17 | dingle | would someone kindly tell me which provider is better? teliax or voicepulse |
| 19:12.11 | Penguin | What is your definition of better? |
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| 19:13.11 | dingle | quality, support, features, pricing |
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| 19:20.13 | dingle | i guess i get no love here :/ |
| 19:20.22 | kleszcz | hmm |
| 19:20.31 | dingle | hi kleszcz |
| 19:20.42 | kleszcz | hi |
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| 19:22.14 | ghost75 | can be seen in log if it is t.38 ? http://pastebin.com/rBpsLPDu |
| 19:26.44 | [TK]D-Fender | ghost75: You clearly aren't bothing to look at SIP DEBUG at all.... |
| 19:26.57 | ghost75 | true |
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| 19:46.10 | dingle | ghost75: what provider do you use? |
| 19:46.27 | ghost75 | see log |
| 19:47.39 | Penguin | seelog.com? |
| 19:48.15 | ghost75 | lookinlogtosee.com |
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| 19:57.13 | vomit | Hello! I have a question in regards to Cisco-CP7940G firmware updating. Currently P0S3-06-3-00 and want to update to P0S3-07-0-00. According to Cisco site, it is allowed and I am doing this correctly although the issue is where the phone is requesting P0S3-07-0-00.sbn where the P0S3-07-3-00 doesn't have a .sbn file. I only have .sb2 and .loads as well as for P0S3-07-1-00 any any higher. Any suggestions? Any input is appreciated! |
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| 20:06.57 | Katty | infobot: seen hmmhesays |
| 20:07.05 | infobot | i haven't seen 'hmmhesays', Katty |
| 20:07.10 | Katty | what?! |
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| 20:13.13 | leifmadsen | not enough m's? :) |
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| 20:16.41 | Penguin | |
| 20:21.47 | dingle | hi Penguin |
| 20:21.58 | dingle | i remember you from #linux |
| 20:22.23 | Penguin | *shrug* |
| 20:22.25 | Penguin | hi |
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| 20:51.35 | BeachBall | can i call into asterisk with a video conference unit and use it like a bridge? |
| 21:12.17 | [TK]D-Fender | yes as long as all sides use the same codec and * supports it in passthrough |
| 21:12.36 | [TK]D-Fender | The is no mixing and there is no transcoding. |
| 21:12.42 | Penguin | He's not here. |
| 21:12.52 | [TK]D-Fender | That too |
| 21:13.03 | [TK]D-Fender | Waited a whole 5 minutes too |
| 21:13.26 | Penguin | I heard that time is money... |
| 21:13.46 | Penguin | I guess that's right. I use up all my time and I have no money. |
| 21:16.01 | Kobaz | and girls = -time + -money |
| 21:18.09 | Kobaz | so life = -time + money - girls which means life = -time + money + -time + -money, which reduces down to life = -2xtime and no money |
| 21:26.39 | [TK]D-Fender | And.... checkout time, heading out.... |
| 21:26.55 | navaismo | seriously nobody in the USA knows "Beakman's World"??? |
| 21:27.30 | Kobaz | transcoding video realtime is going to be fun thing for future asterisk |
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| 21:32.09 | pabelanger | wound't want asterisk to handle it |
| 21:32.20 | pabelanger | maybe something external, but not asterisk |
| 21:32.49 | issackelly | Is there any way I can have different timeouts with the beginning and end of record? |
| 21:32.50 | issackelly | Like, I want to wait for up to 30 seconds before someone starts talking, but once they stop, I don't want to wait for another 10 seconds of silence to end the recording. I thought I could maybe place a BackgroundDetect before a Record function, but that seems like it would cut off the first part of what they say. |
| 21:32.55 | Kobaz | what's wrong with having asterisk do it... it does audio transcoding |
| 21:33.14 | pabelanger | scaling |
| 21:34.20 | Kobaz | more cores? :) |
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| 21:35.02 | pabelanger | Try, but why build it into asterisk when you could possibly leverage something external. |
| 21:35.16 | pabelanger | not saying it exists, but think RTPProxy but for video |
| 21:35.36 | Kobaz | well yeah if you can use an already-built system, by all means |
| 21:35.49 | Kobaz | i wasn't even talking about building one from scratch, there's plenty of libraries and tools |
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| 22:02.09 | darkdrgn2k3 | are tehre any big disadvantages doign VOIP over openVPN is i QOS th openVPN traffic |
| 22:24.35 | Penguin | Again, in English? |
| 22:27.20 | Penguin | I was with you where you asked if there are any big disadvantages to using VoIP over a VPN, but you lost me immediately after that. |
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| 22:33.13 | Kobaz | heh |
| 22:34.04 | Kobaz | darkbasic: any little bit of jitter on the low level transport is going to be bad for the inner vpn traffic |
| 22:34.20 | Kobaz | in general it's okay, but if you have packet delivery issues, you'll really notice it |
| 22:35.01 | WIMPy | How does using a VPN make a difference on that topic? |
| 22:35.36 | Penguin | You have to consider it. |
| 22:35.57 | Nugget | wwhhaatt ddooeess aa jjiitteerrbbuuffffeerr ddoo?? |
| 22:36.08 | Kobaz | because dropping one vpn packet will drop multiple voip packets inside it |
| 22:36.18 | Kobaz | especially if you're using compression |
| 22:36.35 | WIMPy | That's now what I see. |
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| 22:36.50 | WIMPy | But I'm not using compression. |
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