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| 09:23.24 | conosj | I used extension 701 and the password of that extension to login into the UCP, but can't .... how can I login in to UCP ? |
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| 15:47.16 | Lep2 | can this be run on a pi? |
| 15:47.39 | WIMPy | Sure |
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| 17:24.01 | steelbrain | Does anyone if asterisk cdr still works if we drop a column on the table? |
| 17:24.33 | steelbrain | Does it ignore that column and insert the other things or does it not work at all? |
| 17:33.45 | *** join/#asterisk youtmon (~textual@c-73-46-212-142.hsd1.fl.comcast.net) |
| 17:34.12 | youtmon | gm folks |
| 17:34.18 | youtmon | i have a setup with GV, i'm able to make and receive calls when I'm a firewall but not when I'm on my providers cell network. port forwarding is working (i see the packets arriving over udp 5060) when I try to register the phone. I also have RTP ports fwd too, ones mentioned in docs. what else could it be why its not working? |
| 17:37.07 | *** join/#asterisk pjm (~pjm@uhfsatcom.plus.com) |
| 17:40.30 | [TK]D-Fender | <PROTECTED> |
| 17:41.01 | [TK]D-Fender | reword your entire description and provide proper details about how things are actually entworked and show us a failed attempt from CLI |
| 17:41.13 | MaliutaLap | [TK]D-Fender: are you an Archer fan? |
| 17:41.27 | [TK]D-Fender | LAAAAANNNAAAAAAAA |
| 17:41.28 | MaliutaLap | [TK]D-Fender: are we not doing phrasing anymore? |
| 17:41.41 | MaliutaLap | [TK]D-Fender: DANGER ZONE! |
| 17:42.51 | MaliutaLap | https://www.youtube.com/watch?v=hyLWrKh2fB0 |
| 17:44.50 | [TK]D-Fender | <PROTECTED> |
| 17:49.31 | youtmon | simply put, anything accessing asterisk behind the NAT is failing |
| 17:50.41 | [TK]D-Fender | "failing" is not telling us anything useful |
| 17:51.16 | [TK]D-Fender | Show us the failed attempts |
| 17:51.29 | [TK]D-Fender | ~pb |
| 17:51.30 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:51.32 | [TK]D-Fender | ^^^ |
| 17:57.45 | youtmon | Wireshark shows UDP connection over 5060 from the mobile phone to asterisk server. So i know Port fwding is working How do I debug the connection on asterisk server to see if the packet makes it all the way through? |
| 18:03.58 | [TK]D-Fender | You aren't already looking at * CLI with SIP debug enabled? |
| 18:04.07 | [TK]D-Fender | Haven't checked your local firewall? |
| 18:05.21 | youtmon | just turned it on |
| 18:05.52 | youtmon | firewall logs shows a connection from my mobile phone (bria) |
| 18:06.15 | youtmon | enabled sip debug I do not see a hit... |
| 18:06.27 | youtmon | firewall is not enable on asterisk |
| 18:07.10 | youtmon | <PROTECTED> |
| 18:07.32 | [TK]D-Fender | Asterisk doesn't have a firewall |
| 18:07.43 | [TK]D-Fender | And talking about what it working doesn't tell us anything useful |
| 18:07.52 | [TK]D-Fender | SHOW US PACKETS ARE ARRIVING AT ALL\ |
| 18:08.13 | [TK]D-Fender | We also don't care what the client thinks |
| 18:08.17 | [TK]D-Fender | Prove packets ARRIVE |
| 18:08.21 | youtmon | wireshark show packets arriving |
| 18:08.39 | [TK]D-Fender | Where's the firewall dump? |
| 18:08.56 | [TK]D-Fender | Proof that the channel driver is bound to the right port? |
| 18:09.03 | [TK]D-Fender | Show us something useful |
| 18:12.55 | MaliutaLap | [TK]D-Fender: phrasing |
| 18:16.00 | youtmon | SIP debug is only functional when a call is established, does not generate logs from a UDP port connection level |
| 18:18.33 | [TK]D-Fender | Incorrect |
| 18:18.54 | [TK]D-Fender | SIP = call negotiation |
| 18:19.01 | [TK]D-Fender | Every single comm that takes place |
| 18:19.04 | [TK]D-Fender | call or otherwise |
| 18:19.14 | [TK]D-Fender | no packets = nothing hitting your server |
| 18:19.25 | [TK]D-Fender | Prove they are |
| 18:19.27 | [TK]D-Fender | ]Show us |
| 18:19.34 | [TK]D-Fender | Do not tell us "I see stuff" |
| 18:20.03 | [TK]D-Fender | phone registering = sip packets that can be seen in debug |
| 18:20.13 | [TK]D-Fender | subscribing to mailbox or presence? same |
| 18:20.38 | [TK]D-Fender | Actual call attempt inbound? same |
| 18:23.19 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 18:29.59 | *** join/#asterisk youtmon (~yout@c-73-46-212-142.hsd1.fl.comcast.net) |
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| 18:31.54 | MaliutaLap | hands [TK]D-Fender a bigger cluebat |
| 18:32.26 | MaliutaLap | and we start with .... |
| 18:32.29 | MaliutaLap | ~sipnat |
| 18:32.30 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 18:34.15 | [TK]D-Fender | <youtmon> SIP debug is only functional when a call is established, does not generate logs from a UDP port connection level |
| 18:34.22 | [TK]D-Fender | <[TK]D-Fender> Incorrect |
| 18:34.45 | [TK]D-Fender | SIP debug dumps the full contents of whatever packets hit your server |
| 18:35.06 | [TK]D-Fender | Calls, subscriptions, registrations, notify, etc are all packets |
| 18:35.13 | [TK]D-Fender | and UDP is stateless. |
| 18:35.22 | [TK]D-Fender | Using the term "connection" is nonsense |
| 18:35.30 | [TK]D-Fender | Dumb packets arrive or they don't |
| 18:35.55 | youtmon | UDP is connectionless protocol.. how do log connections from debugs ? |
| 18:36.15 | youtmon | tell me from asterisk server and I'll enabel the commend |
| 18:37.11 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 18:39.46 | [TK]D-Fender | You just said it doesn't show anything.. and now you're saying you don't even know how to tro ENABLE it? |
| 18:39.55 | [TK]D-Fender | Birds can't fly... |
| 18:39.57 | [TK]D-Fender | What's a bird? |
| 18:40.19 | [TK]D-Fender | What CHANNEL DRIVER are you using for it? |
| 18:42.52 | youtmon | being that you are the one disputing my finding what is the CLI commands you use to see UDP connections? |
| 18:44.37 | [TK]D-Fender | <[TK]D-Fender> What CHANNEL DRIVER are you using for it? <- |
| 18:45.12 | [TK]D-Fender | As of * 12 there are 2 completely different SIP stacks. |
| 18:45.32 | youtmon | GV |
| 18:46.15 | [TK]D-Fender | GV was XMPP, not "sip" |
| 18:46.38 | [TK]D-Fender | And you've been talking about bria which is clearly not GV |
| 18:46.42 | [TK]D-Fender | Get your story straight |
| 18:47.05 | [TK]D-Fender | <youtmon> Wireshark shows UDP connection over 5060 from the mobile phone to asterisk server. So i know Port fwding is working How do I debug the connection on asterisk server to see if the packet makes it all the way through? |
| 18:47.11 | [TK]D-Fender | 5060 = SIP |
| 18:47.32 | [TK]D-Fender | Since when is a "mobile phone" talking GV to Asterisk? |
| 18:48.24 | youtmon | SIP is the transport used for the setup,,, how many of these have you done besides acting like a smart ass |
| 18:48.56 | [TK]D-Fender | [TK]D-Fender> <[TK]D-Fender> What CHANNEL DRIVER are you using for it? <- |
| 18:48.56 | [TK]D-Fender | <[TK]D-Fender> As of * 12 there are 2 completely different SIP stacks.\ |
| 18:48.57 | youtmon | if you dont have anything worthy to contribute then stay quiet please |
| 18:49.31 | [TK]D-Fender | WHICH CHANNEL DRIVER ARE YOU USING FOR SIP |
| 18:49.42 | [TK]D-Fender | Beccause the debug command DEPNDS on that answer] |
| 18:50.17 | youtmon | pjsip |
| 18:50.42 | [TK]D-Fender | "pjsip set logger on" |
| 18:51.08 | [TK]D-Fender | While you're at it prove that it is even bound to 5060 |
| 18:51.27 | [TK]D-Fender | and not chan_sip |
| 18:52.23 | youtmon | COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME |
| 18:52.23 | youtmon | asterisk 12583 asterisk 11u IPv4 177674 0t0 UDP *:sip |
| 18:52.35 | MaliutaLap | youtmon: to set you fucking straight [TK]D-Fender is one of the most experienced and knowlegeable guys in here. He's probably logged more hours debugging * setups than you have spent alive |
| 18:52.47 | MaliutaLap | youtmon: and like me he doesn't suffer fools easily |
| 18:53.17 | [TK]D-Fender | That proves ASTERISK is listening |
| 18:53.22 | [TK]D-Fender | not that PJSIP IS THE ONE BOUND TO IT |
| 18:53.42 | [TK]D-Fender | 2 completely separate SIP stacks |
| 18:53.49 | [TK]D-Fender | PROVE which is bound where |
| 19:02.33 | hecatae | yay just installed asterisk13 cert 2, pjsip only, off to play |
| 19:02.36 | youtmon | bindings are found in pjsip.transports.conf |
| 19:03.53 | hecatae | makes note |
| 19:04.15 | [TK]D-Fender | That filename does not inherently mean anything to Asterisk |
| 19:04.34 | [TK]D-Fender | And mentioning a filename is not proof |
| 19:06.25 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.62.60) |
| 19:06.52 | youtmon | from CLI there isnt a process defined that shows binding to UDP:5060 |
| 19:07.26 | *** part/#asterisk DragonAzul (~DragonAzu@187.208.62.60) |
| 19:12.12 | [TK]D-Fender | * will tell you whicjh |
| 19:12.27 | [TK]D-Fender | And configs should tell this very clearly as well |
| 19:12.54 | [TK]D-Fender | But over an hour and a half later we've seen nothing |
| 19:15.38 | [TK]D-Fender | "sip show settings" <- shows chan_sip |
| 19:15.53 | [TK]D-Fender | At least LOOKING at the other half is partial prrof |
| 19:16.01 | [TK]D-Fender | Which is 50% more than we have now |
| 19:18.25 | youtmon | what about in that output tells you bindings? |
| 19:18.53 | [TK]D-Fender | The part whjere it tells me the binding |
| 19:21.40 | youtmon | http://pastebin.com/E77HDrdP |
| 19:21.46 | youtmon | 5061 is used fro SIP |
| 19:22.18 | [TK]D-Fender | This should NOT have taken that long to see |
| 19:22.48 | [TK]D-Fender | youtmon> 5061 is used fro SIP <- not quite |
| 19:23.02 | [TK]D-Fender | 5061 is used for *****CHAN_SIP***** |
| 19:23.13 | [TK]D-Fender | Therefor the other binding logically only leaves PJSIP |
| 19:23.38 | [TK]D-Fender | So PJSIP is what is processing packets on 5060 |
| 19:24.02 | youtmon | which port SIP phones register over 5060 or 5061? |
| 19:24.18 | [TK]D-Fender | Depends on what you told it to use obviously |
| 19:24.31 | [TK]D-Fender | If I tell mine to use 8000 it will |
| 19:24.52 | [TK]D-Fender | If you didn't tell it anything at all, odds are it'll use the standard port |
| 19:25.03 | [TK]D-Fender | Which is common sense |
| 19:25.43 | [TK]D-Fender | Then again.... you have wireshark captures that answered that before you walked in. This shouldn't have needed to be asked at all |
| 19:26.02 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 19:26.22 | *** join/#asterisk sgduser (~sulaiman@82.178.80.83) |
| 19:27.54 | youtmon | WS shows 5060 and I followed the DOC port forwarding suggestions but there are other ports listening on the server. i asked for verification |
| 19:29.37 | [TK]D-Fender | only 2 things should ahve any odds of binding to that port' |
| 19:29.51 | [TK]D-Fender | The 2 "now"-standard SIP channel drivers |
| 19:30.23 | [TK]D-Fender | Where's the debug? |
| 19:30.36 | [TK]D-Fender | I ghave you the command for this a LONG while back |
| 19:30.50 | [TK]D-Fender | Where's the firewall dump showing it isn't being BLOCKED at that level? |
| 19:33.19 | youtmon | the debug command I enabled |
| 19:33.26 | [TK]D-Fender | SHOW US |
| 19:33.33 | [TK]D-Fender | What part of "show" are you having trouble with? |
| 19:33.39 | [TK]D-Fender | Where is the FIREWALL DUMP? |
| 19:33.56 | [TK]D-Fender | PASTEBIN <----------- |
| 19:33.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 19:34.27 | [TK]D-Fender | I don''t SEE that you did it right. I don't SEE a firewall dump showing that you are screwing yourself. |
| 19:34.29 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 19:34.37 | [TK]D-Fender | We are wasting HOURS and you show nothing |
| 19:35.10 | youtmon | 14:34:29.219380 IP mobile-166-172-188-038.mycingular.net.facilityview > 10.111.112.101.sip: SIP, length: 639 |
| 19:35.20 | youtmon | there is the connection from my mobile |
| 19:35.24 | [TK]D-Fender | That is external garbage |
| 19:35.33 | [TK]D-Fender | ASTERISK CLI |
| 19:35.40 | youtmon | its command used was tcpdump udp port 5060 |
| 19:35.40 | [TK]D-Fender | FIREWALL DUMP |
| 19:35.46 | [TK]D-Fender | NO |
| 19:35.56 | [TK]D-Fender | ASTERISK PJSIP SET LOGGER ON |
| 19:36.09 | [TK]D-Fender | ENABLE DEBUG LIKE I TOLD YOU |
| 19:36.12 | [TK]D-Fender | Aand SHOW US |
| 19:36.25 | [TK]D-Fender | And show us the damn firewall dump |
| 19:36.30 | [TK]D-Fender | and stop wasting our time |
| 19:36.48 | youtmon | ok mister.. lol |
| 19:37.20 | youtmon | logger enabled .... no logs generated yet |
| 19:37.55 | [TK]D-Fender | <[TK]D-Fender> FIREWALL DUMP <------------------------ |
| 19:37.57 | [TK]D-Fender | Where is it>? |
| 19:37.58 | youtmon | trying phone registration from 4G |
| 19:38.05 | [TK]D-Fender | I4 G means nothing |
| 19:38.13 | youtmon | i see no log entries posted |
| 19:38.14 | [TK]D-Fender | FIREWALL DUMP. |
| 19:38.15 | [TK]D-Fender | NOW |
| 19:38.56 | [TK]D-Fender | youtmon> i see no log entries posted <- what "log entries"? What are you even calling a "log"? |
| 19:38.58 | youtmon | pardon... what command you gave me for a firewall dump? |
| 19:39.06 | [TK]D-Fender | IPTABLES <- |
| 19:39.10 | [TK]D-Fender | facepalms |
| 19:39.10 | youtmon | log to screen |
| 19:39.19 | [TK]D-Fender | iptables --list |
| 19:39.21 | [TK]D-Fender | ^ |
| 19:40.51 | youtmon | http://pastebin.com/FJwp2z68 |
| 19:41.38 | [TK]D-Fender | trash it all |
| 19:41.43 | [TK]D-Fender | iptables --flush |
| 19:43.04 | *** join/#asterisk TeslaZap (~teslazap@unaffiliated/teslazap) |
| 19:43.13 | youtmon | unable to SSH into box |
| 19:43.25 | [TK]D-Fender | ... |
| 19:43.31 | [TK]D-Fender | Chain INPUT (policy DROP) |
| 19:43.32 | [TK]D-Fender | GAH |
| 19:43.37 | [TK]D-Fender | stupid firewall |
| 19:43.52 | youtmon | lol |
| 19:44.01 | youtmon | trying console |
| 19:44.30 | [TK]D-Fender | Go fix this to properly allow all so you don't have it screwing you |
| 19:44.36 | [TK]D-Fender | heads out for a while |
| 19:56.20 | youtmon | [TK]D-Fender WOW... all working!! |
| 19:56.34 | youtmon | friggin liunx FW was filtering crap |
| 20:01.14 | youtmon | [TK]D-Fender thanks for your help in spite of... lol |
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| 21:14.50 | PipBoy3k | Hi guys, its been a long time since I've used asterisk in production (1.4 i think?). Does asterisk still rely on the processor for timing? or has something like a monotonic clock been introduced? |
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| 22:32.05 | craigify | I love reading the history of this channel :) |
| 22:40.46 | MaliutaLap | craigify: phrasing |
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| 23:15.32 | *** join/#asterisk Lope (~Lope@192.228.141.117) |
| 23:16.05 | Lope | Is there a way to make 2 softphones connected on a call change codecs during the call? |
| 23:16.17 | Lope | I'd like to test different codecs |
| 23:18.58 | [TK]D-Fender | no |
| 23:19.14 | MaliutaLap | test them with different calls |
| 23:19.50 | Lope | okay, what codec gives the best quality? |
| 23:20.52 | youtmon | 711 |
| 23:21.08 | Lope | in sip.conf for both of my softphones I have a line that says allow=!all,adpcm,a_mu,g726,lpc10,speex,alaw,g722,gsm,resample,ulaw |
| 23:21.22 | Lope | are those all the codecs available with asterisk? |
| 23:21.29 | MaliutaLap | I use g729 |
| 23:21.38 | youtmon | ulaw |
| 23:23.53 | Lope | I tried allow=!all,g729 and now my 2 softphones can't call. It says "not acceptable here" |
| 23:23.57 | youtmon | g729 is bandwidth friendly not best signal quality @ 8K |
| 23:24.01 | *** join/#asterisk rootshell (~root@71-9-117-244.static.lsan.ca.charter.com) |
| 23:24.36 | Lope | i don't have a module with g729 in it's name, I only have a module with g722 in it's name? |
| 23:25.03 | youtmon | you may have to set g729 under the SIP conf |
| 23:26.05 | lvlinux | Lope g729 is a paid add-on codec. Unless you specifically bought it and installed it, you probably don't have it. |
| 23:26.06 | Lope | I entered allow=!all,g729 under both softphone configs in sip.conf |
| 23:26.17 | Lope | okay, then I don't have it :) |
| 23:26.24 | Lope | Is g722 any good? |
| 23:26.31 | lvlinux | I like g722 |
| 23:26.42 | lvlinux | it's a well supported wideband codec |
| 23:26.44 | youtmon | is this on your LAN? |
| 23:26.56 | lvlinux | ^^^ good point |
| 23:27.06 | youtmon | if your LAN is 100+mbps i'd stick with ulaw |
| 23:27.17 | lvlinux | g722 is better than ulaw |
| 23:27.25 | lvlinux | sounds much much better |
| 23:27.47 | lvlinux | but g722 won't be supported by most ITSPs |
| 23:28.07 | Lope | When I used allow=!all,ulaw then it connected with PCM8000 and the quality wasn't the greatest. when I use g722 it uses g722 at a 16000 sample rate, and sounds good. |
| 23:28.23 | Lope | What is an ITSP? |
| 23:28.29 | lvlinux | VoIP provider |
| 23:29.01 | Lope | Oh I see. Can you guys recommend where to get cheap international DIDs? I'm interested in malaysia, Thailand, Singapore, South Africa |
| 23:29.05 | youtmon | sampling rate is crazy high for g722 |
| 23:29.13 | lvlinux | in other words, if you are making calls ot the outside world, the only thing that will get you anywhere is ULAW or ALAW, and most support 729 and a few gsm. |
| 23:29.22 | WIMPy | Which ITSP would care about the CODEC unless you want tp call the PSTN? |
| 23:29.23 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 23:29.36 | Lope | how much does it cost to buy g729? |
| 23:29.42 | WIMPy | youtmon: It's only double. |
| 23:29.52 | Lope | is it possible to just copy g729 to test it? |
| 23:30.06 | WIMPy | Lope: IIRC it's 10 US$/channel. |
| 23:30.18 | lvlinux | Lope, there is a "free" g729 library out there, but people have mixed success with it. |
| 23:30.39 | lvlinux | g729 quality is slightly inferior to ulaw, but at a much smaller bandwidth |
| 23:30.44 | Lope | lvlinux: and by free, do you mean pirated or reverse engineered? |
| 23:31.00 | Lope | oh, bugger that, I'll just use ulaw then. |
| 23:31.03 | lvlinux | reverse engineered I believe |
| 23:31.07 | Lope | Nothing worse than bad quality VoIP. |
| 23:31.18 | lvlinux | but it's not technically legal, depending on your point of view |
| 23:31.19 | Lope | lvlinux: do you have a link? |
| 23:31.25 | WIMPy | You usually get kicked out of the channel for mentioning the "free" version. |
| 23:31.49 | lvlinux | well that's why I put it in quotes :-) |
| 23:32.07 | lvlinux | it's "freely available" as in torrented movies lol |
| 23:32.36 | Lope | what's better quality, ulaw or alaw? |
| 23:33.18 | youtmon | "better" is subjective to each situation |
| 23:33.19 | Lope | What are all the spam calls I was getting when I had a "catchall" extension number? I was getting calls to extensins like 101 411 and a bunch of other 3 and 4 digit crap? |
| 23:33.50 | Lope | How do you guys keep spam out of your asterisk? do you have a IP whitelist for your DID providers? |
| 23:33.53 | lvlinux | Lope ulaw and alaw are identical quality. It's just some bit ordering that is different. |
| 23:34.18 | lvlinux | ulaw is used primarily in the US (and maybe Japan???) and Alaw everywhere else. |
| 23:35.13 | lvlinux | Lope if you need a low bandwidth codec, I like ilbc. And speex is good too (it has some nice features). |
| 23:36.05 | lvlinux | Lope g722 is 64kbps, same as ulaw/alaw. |
| 23:36.42 | Lope | ah great, thanks for the recommendations :) |
| 23:37.01 | lvlinux | http://kb.kerio.com/product/kerio-operator/server-configuration-kerio-operator/bandwidth-used-by-the-different-codecs-1107.html |
| 23:37.06 | lvlinux | look there ^^^ |
| 23:38.14 | Lope | what about g723? |
| 23:38.20 | lvlinux | Lope, you can set your mobile phone softphone to use a codec with low bandwidth like ilbc or speex, and use g722 in the house/office where you have more bandwidth available. |
| 23:38.28 | lvlinux | g723 is kindof old IIRC |
| 23:39.09 | WIMPy | Yes, G.723 is superseededbyt G.726. |
| 23:39.23 | lvlinux | GSM is a goto low bandwidth codec too. But sounds nasty to me. |
| 23:39.44 | Lope | I didn't see a module with g726 in it's name in my asterisk installation, is it a paid codec? |
| 23:39.47 | lvlinux | Ah ddn't know that WIMPy. Good to know. |
| 23:39.51 | WIMPy | There isn't _one_ GSM CODEC. |
| 23:40.02 | lvlinux | true ^ |
| 23:40.17 | lvlinux | g726 isn't paid, it's probably in there. |
| 23:40.45 | rootshell | anyone insterested in a agi script to launch nmap -sv -P0 -Pn scans from your phone? |
| 23:40.57 | rootshell | and have output read via TTS |
| 23:41.01 | WIMPy | For true (old fashion) GSM it's three CODECS. The worst of which (from the 80s) is available in VOIP land. |
| 23:41.39 | WIMPy | Then we get additional ones since UMTS. |
| 23:42.27 | lvlinux | ah |
| 23:44.48 | lvlinux | rootshell: sounds neat, but what exactly would be the useful purpose of launching nmap scans from your phone? |
| 23:46.50 | [TK]D-Fender | Asterisk does not directly support G.723 and it is BAD quality comparitively |
| 23:47.25 | [TK]D-Fender | <lvlinux> g729 quality is slightly inferior to ulaw, but at a much smaller bandwidth <- more than just slightly, but "good enough" for most usage" |
| 23:48.09 | [TK]D-Fender | <Lope> I tried allow=!all,g729 and now my 2 softphones can't call. It says "not acceptable here" <- because it didn't match the other side |
| 23:48.25 | WIMPy | For decent quality wit less BW, I'd look at G.726. |
| 23:48.27 | [TK]D-Fender | And nobody uses G.726 |
| 23:49.04 | youtmon | [TK]D-Fender which low bandwidth codec is recommended that is asterisk supported and free |
| 23:49.25 | WIMPy | It's in extremely wide use. Every cordless phone uses G.726. |
| 23:49.31 | youtmon | that nmap command isnt correct |
| 23:49.34 | [TK]D-Fender | How despeare are you for bandwidth? |
| 23:49.39 | [TK]D-Fender | desperate* |
| 23:49.42 | WIMPy | Unless you're lucky and it does G.722. |
| 23:51.10 | youtmon | i like to play it safe and stick to widely supported standards like g729 (not free) and g711 |
| 23:51.13 | [TK]D-Fender | <Lope> How do you guys keep spam out of your asterisk? do you have a IP whitelist for your DID providers? <- whitelist IP's, track failures and firewall them out, etc |
| 23:51.36 | [TK]D-Fender | Use VPN exclusively, etc |
| 23:51.46 | [TK]D-Fender | All the basic things everyone should think of. |
| 23:51.48 | youtmon | ok whitelist at the perimeter firewall |
| 23:51.57 | Lope | lvlinux: I don't have ilibc in my asterisk install and speex doesn't seem to be supported by SFLphone (a linux softphone). but gsm works for me. |
| 23:52.03 | WIMPy | As has been pointed out, speex is nice, but not that widely supported. |
| 23:52.45 | Lope | can you guys recommend a FOSS linux softphone? |
| 23:53.23 | youtmon | is there a doc that explains dial plans if one has more than one GV accounts? |
| 23:53.40 | youtmon | i like BRIA |
| 23:53.52 | lvlinux | Lope: yate-qt4, twinkle, pjsua :-) |
| 23:54.08 | [TK]D-Fender | <youtmon> is there a doc that explains dial plans if one has more than one GV accounts? <- that isn't a "thinkg" |
| 23:54.13 | [TK]D-Fender | Dialplan = all call processing. |
| 23:54.17 | [TK]D-Fender | It does whatever you tell it to |
| 23:54.21 | [TK]D-Fender | Dialplan = programming |
| 23:54.42 | [TK]D-Fender | Ther is no such thing as "guideline", or "default" or basic". There is only what you make for your own needs |
| 23:55.00 | [TK]D-Fender | Having 2 accounts doesn't mean anything specific to how you choose to arrange your dialplan |
| 23:55.01 | lvlinux | I think he wants a tutorial |
| 23:55.09 | lvlinux | But I don't know of any that fit what he wants. |
| 23:55.10 | Lope | anyone know what codec is supported by the a_mu module? |
| 23:55.21 | [TK]D-Fender | The question was asked in relation to "having 2 GV accounts". |
| 23:55.25 | [TK]D-Fender | That very factor means nothing |
| 23:55.49 | lvlinux | ok so he meant "using 2 GV accounts to feed calls through *" |
| 23:56.01 | WIMPy | Lope: G.711 |
| 23:56.03 | Lope | anyone know what codec is supported by the lcp10 module? |
| 23:56.12 | youtmon | how about which profiles to use based on area code? |
| 23:56.14 | [TK]D-Fender | I think we should throw conjecture out the door untill a meaninful question is actually asked |
| 23:56.30 | Lope | WIMPy: thanks. Is G.711 worth using? |
| 23:56.32 | [TK]D-Fender | You decide what you dial |
| 23:56.36 | lvlinux | youtmon: hwat is a "profile" |
| 23:56.37 | [TK]D-Fender | as I said dialplan = PROGRAMMING |
| 23:56.51 | lvlinux | Lope: g711 is the defacto standard for PSTN |
| 23:56.59 | [TK]D-Fender | YOU look up whatever you want to look up using whatever lists sources, etc you have and make your call do whatever you want based on that |
| 23:57.03 | WIMPy | Lope: That's th one that's used in the PSTN. The one that's always supported, everywhere. |
| 23:57.28 | lvlinux | Lope: it is as good quality as you can get at 8000hz uncompressed |
| 23:57.48 | WIMPy | Well, it has reduction. |
| 23:58.04 | [TK]D-Fender | "companding", not compression. |
| 23:58.11 | lvlinux | yes |
| 23:58.13 | WIMPy | It's 12/13 Bit samples reduced to 8 bit. |
| 23:58.14 | [TK]D-Fender | but it's the baseline |
| 23:58.18 | Lope | weird. I set allow=!all,g711 and neither SFLphone nor my android softphone support it? |
| 23:58.27 | Lope | "Not acceptable here" whatever that means. |
| 23:58.32 | [TK]D-Fender | we don't see what you told it to allow2 |
| 23:58.32 | WIMPy | Because it's not a valid name. |
| 23:58.36 | lvlinux | ulaw |
| 23:58.54 | WIMPy | G.711 is called alaw or ulaw. Chose your flavour. |
| 23:58.59 | [TK]D-Fender | getting the names right would be nice as well |
| 23:59.03 | Lope | WIMPy: ohhh, I see :) |
| 23:59.33 | Lope | yet G.722 is called g722. They make it so intuitive :p |
| 23:59.37 | WIMPy | Where ulaw actually means ยต-law. |
| 23:59.50 | lvlinux | well there is only one flavor of g722 IIRC |