| 00:16.27 | *** join/#asterisk klow (~textual@c-67-161-106-89.hsd1.wa.comcast.net) |
| 00:44.42 | *** join/#asterisk adeeln (~adeel@CPE00505698b193-CM0c473dd4c8c0.cpe.net.cable.rogers.com) |
| 00:48.04 | xochilpili | [TK]D-Fender, this is the config and the sip debug in both servers |
| 00:48.06 | xochilpili | http://pastebin.com/QnJBBf9h |
| 00:50.08 | [TK]D-Fender | Retransmitting #2 (no NAT) to 205.186.175.114:5060: |
| 00:50.13 | [TK]D-Fender | tha is a PUBLIC IP |
| 00:50.26 | [TK]D-Fender | And you say your server B is a PRIVATE one. |
| 00:50.32 | [TK]D-Fender | So why are you sending to the PUBLIC IP? |
| 00:51.44 | [TK]D-Fender | You are also showing only the debug & config from ONE side |
| 00:53.32 | xochilpili | [TK]D-Fender, no, im showing both servers |
| 00:54.00 | xochilpili | ============= SERVER B: 10.0.253.7 ================ << |
| 00:54.30 | [TK]D-Fender | <[TK]D-Fender> Retransmitting #2 (no NAT) to 205.186.175.114:5060: <------------------ |
| 00:56.00 | xochilpili | [TK]D-Fender, i dont know why is calling this IP, is not mine :/ |
| 00:56.16 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 00:56.25 | [TK]D-Fender | <PROTECTED> |
| 00:56.34 | [TK]D-Fender | Reliably Transmitting (no NAT) to 205.186.175.114:5060: |
| 00:56.34 | [TK]D-Fender | INVITE sip:1100@server_b SIP/2.0 |
| 00:56.46 | [TK]D-Fender | Very clear about where * was told to contact them at |
| 00:57.13 | xochilpili | [TK]D-Fender, oh! if i ping to server_b i got that ip address |
| 00:58.04 | [TK]D-Fender | but that's NOT the IP you said |
| 00:58.20 | [TK]D-Fender | So why should we be seeing that? |
| 00:58.26 | xochilpili | no, it's not my ipaddress |
| 00:58.34 | xochilpili | i dont know what server is that |
| 00:58.49 | xochilpili | but i dont know either why is calling to that server |
| 00:59.00 | [TK]D-Fender | Because something REGISTERED to it |
| 00:59.26 | [TK]D-Fender | [server_b] host=dynamic |
| 00:59.45 | [TK]D-Fender | You have FIXED IP's. Why are you even REGISTERING in the first place? |
| 00:59.55 | [TK]D-Fender | You register when you ar MOBILE. |
| 01:00.08 | [TK]D-Fender | IIf your IP is expected to change |
| 01:00.29 | xochilpili | [server_b] host=10.0.253.7 make sense? |
| 01:00.36 | [TK]D-Fender | Is that hte IP? |
| 01:00.44 | [TK]D-Fender | then that is what you should be |
| 01:01.01 | xochilpili | yes, server A is 10.0.253.6 while server_B is 10.0.253.7 |
| 01:03.08 | xochilpili | i got this changing host chan_sip.c:17154 register_verify: Peer 'server_a' is trying to register, but not configured as host=dynamic |
| 01:04.23 | *** join/#asterisk jjrh (~jjrh@2607:f0b0:8:801c:d87:100c:9d70:4ab6) |
| 01:04.39 | [TK]D-Fender | Now the you've set hosts... STOP TRYING TO REGISTER |
| 01:05.26 | xochilpili | for both servers on each ? |
| 01:05.46 | xochilpili | i mean in hosts i need to add both: server_a and server_b on each server right? |
| 01:06.21 | [TK]D-Fender | No |
| 01:06.31 | [TK]D-Fender | ! account that MATCHES on both sides |
| 01:06.33 | [TK]D-Fender | 1 |
| 01:06.38 | [TK]D-Fender | NO REGISTERING |
| 01:06.49 | [TK]D-Fender | Each side has the OTHER SIDES IP on the host |
| 01:08.08 | *** join/#asterisk KerioMorgan (~Adium@us.kerio.com) |
| 01:10.53 | xochilpili | i dont get : aacount that matches on both sides ? |
| 01:11.18 | [TK]D-Fender | 1 peer on each side |
| 01:11.19 | xochilpili | both accounts server_a and server_b are registered in both sides |
| 01:11.25 | [TK]D-Fender | NO REGISTERIONG. |
| 01:11.34 | [TK]D-Fender | What part are you having trouble understanding? |
| 01:11.52 | [TK]D-Fender | NO REGISTER AT ALL. REMOVE THEM. ****NOW**** |
| 01:12.05 | xochilpili | remove register=> ?? |
| 01:12.10 | [TK]D-Fender | You KNOW the IP on the other side. Set them in the peer you use on each isde |
| 01:12.15 | [TK]D-Fender | NO REGISTER AT ALL. REMOVE THEM. ****NOW**** |
| 01:12.35 | xochilpili | [TK]D-Fender, i also have an englsh issues :/ |
| 01:12.44 | xochilpili | please dont be impatient |
| 01:12.55 | [TK]D-Fender | <[TK]D-Fender> NO REGISTERING |
| 01:12.56 | xochilpili | i have remove register=> in both sides |
| 01:13.00 | [TK]D-Fender | [TK]D-Fender> Now the you've set hosts... STOP TRYING TO REGISTER |
| 01:14.16 | xochilpili | hosts > http://pastebin.com/AcZBeDhc |
| 01:16.55 | [TK]D-Fender | Why are you screwing around iwth the HOSTS file? |
| 01:17.18 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:18.01 | xochilpili | what you mean? |
| 01:18.07 | xochilpili | i not ok ? |
| 01:18.18 | xochilpili | is not ok? |
| 01:18.36 | [TK]D-Fender | NO |
| 01:18.41 | [TK]D-Fender | That is not an Asterisk config |
| 01:18.50 | [TK]D-Fender | whay are you screwing around with OTHER stuff? |
| 01:19.18 | xochilpili | <[TK]D-Fender> Now the you've set hosts... STOP TRYING TO REGISTER << |
| 01:19.25 | [TK]D-Fender | <xochilpili> [server_b] host=10.0.253.7 make sense? |
| 01:19.26 | xochilpili | or you mean host= << ? |
| 01:19.34 | xochilpili | damn |
| 01:19.35 | xochilpili | sorry |
| 01:19.37 | [TK]D-Fender | HOST= <--- IN SIP.CONF |
| 01:19.52 | [TK]D-Fender | NOT DYNAMIC |
| 01:20.03 | xochilpili | sorry, sorry |
| 01:21.36 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 01:22.29 | xochilpili | [TK]D-Fender, but for example. in sip.conf in the server_a host = this is the IPADDR of SERVER A or SERVER B??? |
| 01:23.09 | [TK]D-Fender | THE OTHER SIDE] |
| 01:23.36 | [TK]D-Fender | <[TK]D-Fender> Each side has the OTHER SIDES IP on the host |
| 01:26.02 | xochilpili | [TK]D-Fender, this is the new sip.conf settings : http://pastebin.com/aP4cBCvk |
| 01:29.35 | xochilpili | [TK]D-Fender, http://pastebin.com/9gfFUAtp << debug server A |
| 01:29.48 | xochilpili | http://pastebin.com/caSH8fSE << debug server B |
| 01:30.30 | xochilpili | in server b there's a getaddrinfo("server_a", "(null)", ...): Name or service not known ??? |
| 01:30.43 | xochilpili | and server A do anything |
| 01:31.14 | [TK]D-Fender | You're still using a named host |
| 01:31.28 | [TK]D-Fender | it's doing a DNS lookup beacuse you stil have a reference fkloating around |
| 01:32.51 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:6355:b61:14e7:dbba) |
| 01:33.42 | xochilpili | [TK]D-Fender, where to change this? i have not enabled dnsmgr.conf |
| 01:34.26 | xochilpili | srvlookup=no << |
| 01:34.50 | [TK]D-Fender | ... |
| 01:35.24 | [TK]D-Fender | ;register => server_b:p4pr1k4@10.0.253.7/server_a |
| 01:35.27 | [TK]D-Fender | Garbage like this |
| 01:35.54 | [TK]D-Fender | nevermind. |
| 01:35.55 | [TK]D-Fender | wrong line |
| 01:36.03 | [TK]D-Fender | restat both *'s |
| 01:36.19 | xochilpili | done |
| 01:36.23 | xochilpili | restart? |
| 01:36.24 | [TK]D-Fender | and show the configs for both |
| 01:36.27 | [TK]D-Fender | restart * |
| 01:36.31 | [TK]D-Fender | complete |
| 01:36.33 | [TK]D-Fender | not just a reload |
| 01:36.42 | [TK]D-Fender | "core restart now" |
| 01:36.42 | xochilpili | service asterisk restart |
| 01:36.47 | [TK]D-Fender | Or that |
| 01:36.51 | xochilpili | ok done |
| 01:36.52 | *** join/#asterisk wil_syd (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |
| 01:37.07 | xochilpili | ERROR[15711][C-00000000]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("server_a", "(null)", ...): Name or service not known |
| 01:37.12 | xochilpili | i got the same error |
| 01:37.16 | xochilpili | in server b |
| 01:37.37 | [TK]D-Fender | Still using that HOST entry instead of an IP somewhere |
| 01:39.08 | xochilpili | http://pastebin.com/MwCbUt4Z << |
| 01:39.35 | [TK]D-Fender | Go look at what it's trying to do that's referencing it. |
| 01:40.17 | xochilpili | in sip.conf i have change [server_a] host = 10.0.253.7 ; (ip address server B) and in [server_b] host=10.0.253.7 ; (ip address server A) |
| 01:40.28 | [TK]D-Fender | make the peer names match |
| 01:40.35 | [TK]D-Fender | and type=peer |
| 01:40.41 | [TK]D-Fender | not friend |
| 01:40.46 | xochilpili | oks |
| 01:41.42 | xochilpili | erver_a 10.0.253.7 Auto (No) No 5060 Unmonitored << server a : sip show peers |
| 01:42.03 | xochilpili | server_b 10.0.253.6 No No 5060 Unmonitored << server b : sip show peers |
| 01:42.22 | xochilpili | and i got the same: getaddrinfo("server_a", "(null)", ...): Name or service not known |
| 01:42.42 | [TK]D-Fender | You're not shoing the FULL SIP DEBUG to show WHY it's making that attempt in the first place |
| 01:43.32 | xochilpili | http://pastebin.com/11i13iZv << server b DEBUG |
| 01:44.31 | [TK]D-Fender | - Executing [1100@from-internal:2] Dial("SIP/100-00000002", "SIP/server_a/1100") in new stack |
| 01:44.56 | *** join/#asterisk wil_syd2 (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |
| 01:45.00 | [TK]D-Fender | Where do we see you have a peername matching that on that server? |
| 01:46.02 | xochilpili | http://pastebin.com/PWhAM7FS << extensions.conf in server A |
| 01:46.35 | xochilpili | but the call is not reporting on server_a |
| 01:47.00 | [TK]D-Fender | I did not say "extensions.conf" |
| 01:47.03 | [TK]D-Fender | I said PEERNAME |
| 01:47.41 | xochilpili | i remove the register=> !! |
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| 11:42.39 | kchehab | while loading Record command after call is answered , reocrded file .wav playback sound is Chewing , for example call duration is 20 seconds you will find wav file lenght 45 seconds |
| 11:42.41 | kchehab | any idea |
| 11:43.16 | kchehab | btw it was working perfectlyy before , its acting like that from couple of days only |
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| 12:39.06 | nixnothing | https://www.youtube.com/watch?v=eUochCGC9Ag |
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| 12:39.25 | nixnothing | *wrong tab* |
| 12:40.15 | nixnothing | *but still asterisk related, so I guess It works out* |
| 12:40.21 | Chainsaw | Well at least it's vaguely relevant. I think you'll get away with it :) |
| 12:40.31 | nixnothing | XD |
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| 13:40.45 | nixnothing | oh god |
| 13:40.58 | nixnothing | our coffee machine was down today |
| 13:40.59 | file | falls over |
| 13:41.45 | nixnothing | they just fixed it |
| 13:42.30 | TandyUK | celebrate |
| 13:42.35 | TandyUK | assuming it makes Tea that is :P |
| 13:42.43 | nixnothing | https://s-media-cache-ak0.pinimg.com/736x/24/de/ca/24decaddab89339f7950c4a653ab2b83.jpg |
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| 14:44.38 | s8y | hello, I would like to disable modules that are not needed in my infrastructure. Since I don't have cisco phones nor pstn can I disable skinny and dhadi? |
| 14:45.06 | [TK]D-Fender | yes |
| 14:45.24 | ModFather | hi [TK]D-Fender :) |
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| 14:45.39 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:48.29 | s8y | thanks [TK]D-Fender, is it safe to also disable ael? |
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| 14:49.13 | [TK]D-Fender | yes |
| 15:00.08 | s8y | thanks again [TK]D-Fender, would that be noload => func_aes.so inside modules.conf or is it done in some other way? |
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| 15:01.00 | [TK]D-Fender | that'd be the way |
| 15:01.18 | monsterco | Hi everyone - I see that there are few libraries for ARI but seems they are not maintained. Is there an offiial Digium or other well maintained python library for ARI? |
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| 15:04.38 | [TK]D-Fender | https://www.google.ca/#q=python+ari+asterisk |
| 15:04.44 | [TK]D-Fender | VERY easy Google search. |
| 15:04.57 | [TK]D-Fender | Google is your friend. Love the Google. Use the Google. |
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| 15:18.59 | monsterco | I have checked all those available and are old or not maintained...hence my question if there is an official library for this... |
| 15:19.07 | monsterco | so i guess not |
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| 15:36.08 | monsterco | Using Asterisk ARI - can i connect two calls together? is that bridging? |
| 15:38.00 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 15:38.47 | [TK]D-Fender | "Old" |
| 15:38.51 | monsterco | Is this what i need or is there something simpler? https://wiki.asterisk.org/wiki/display/AST/Bridges |
| 15:38.52 | [TK]D-Fender | How old is "old"? |
| 15:39.14 | monsterco | currently, I am using Call Files to make a call and then directing that to dialplan to make the second leg of call |
| 15:39.28 | monsterco | [TK]D-Fender - two years with no questions answered... |
| 15:39.32 | [TK]D-Fender | What does any of that have to do with ARI? |
| 15:39.47 | [TK]D-Fender | <monsterco> [TK]D-Fender - two years with no questions answered... <- link it |
| 15:40.52 | monsterco | ARI python library is OLD |
| 15:41.05 | monsterco | is the relevance - see my original question |
| 15:41.09 | [TK]D-Fender | Link it |
| 15:42.34 | monsterco | https://github.com/asterisk/ari-py/issues |
| 15:43.44 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:43.44 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:43.45 | monsterco | Latest issue answered: Sep 15, 2014 |
| 15:44.52 | [TK]D-Fender | https://www.google.ca/#q=asterisk+ARI+python |
| 15:45.06 | [TK]D-Fender | Well you can eitehr follow the docs and code what you need or Google on. |
| 15:45.09 | [TK]D-Fender | I'm not seeing much |
| 15:45.19 | [TK]D-Fender | Is tehre a specific problem with the aspect you need? |
| 15:45.50 | [TK]D-Fender | And before we even get to this point.. what do you actually need? |
| 15:47.02 | drmessano | I'm not sure why you would need a library |
| 15:47.53 | drmessano | It's REST.. |
| 15:49.00 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 16:00.07 | *** join/#asterisk CrowX- (~CrowX-@79.141.122.86) |
| 16:00.09 | monsterco | drmessano - I was trying to establish that. I was in the process of doing research... |
| 16:00.27 | monsterco | not sure why a library exists and if they are usefull to me or I should simply connect directly.. |
| 16:01.00 | monsterco | side question: "SIP/4154456665-out-00002fc2 is making progress passing it to IAX2/220-6634" <<< is this using Bridge? |
| 16:01.07 | drmessano | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 |
| 16:01.17 | drmessano | Literally its all on the Wiki page for ARI |
| 16:01.23 | drmessano | Which should be the first place you looked |
| 16:01.33 | monsterco | [TK]D-Fender - I am looking to use ARI for many different operations. But to start with, I want to call one party and then set caller ID and call another party and join the calls |
| 16:02.00 | [TK]D-Fender | How is call-files not enough already? |
| 16:02.06 | monsterco | drmessano - that is the page I mentioned I got the info from |
| 16:02.24 | monsterco | [TK]D-Fender - call files work beautifully - but I want to upgrade.... |
| 16:02.47 | monsterco | they can't tell me channel status to show on web though...ARI will help show when a call is established, or hangup etc... |
| 16:03.03 | monsterco | plus it can be controlled from another server end rather than files on asterisk... |
| 16:06.55 | *** join/#asterisk themayor (~themayor@unaffiliated/themayor) |
| 16:08.13 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 16:08.18 | Dovid | Anyone here for Belgium? |
| 16:09.19 | monsterco | are "core show applications" in asterisk CLI same and works for ARI? |
| 16:10.21 | file | no, you can't execute dialplan applications from ARI - unless you send the channel back to the dialplan |
| 16:11.34 | *** join/#asterisk consolejazz (~consoleja@fsf/member/consolejazz) |
| 16:11.45 | monsterco | @file - Thanks. weired....I thought you can do this there....hmmm...can I set CallerID? |
| 16:12.03 | file | when you originate an outgoing call in ARI, yes |
| 16:12.09 | file | ARI is for writing telephony applications |
| 16:12.17 | file | it provides the primitives to do so |
| 16:12.38 | monsterco | @file - so what do I use to originate an outgoing call in ARI? |
| 16:12.43 | file | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 describes it |
| 16:12.51 | monsterco | I thought that is the: Stasis and GET /applications/{applicationName} |
| 16:13.34 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-originate |
| 16:13.50 | file | and that GET is to get details about an ARI application |
| 16:15.44 | consolejazz | Hi. Running Debian headless on pogoplug v4 (low-powered plug computer). Wish to install Asterisk and connect to my PBX trunk (voip.ms) and use an ATA with the setup |
| 16:15.51 | consolejazz | Best of following these instructions, right? https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source |
| 16:16.15 | consolejazz | Considering there's no official, up-to-date Asterisks packages in official Debian package manager.. |
| 16:16.34 | consolejazz | Just looking for basic bare bones setup |
| 16:16.58 | monsterco | @file - thanks, I have been reading those...I have implemented the example extension 1000 which allows me to interject Hello World into the call...so there is no way to use ARI to originate a call? |
| 16:17.12 | file | yes, there is |
| 16:17.21 | monsterco | i mean using like POST |
| 16:17.30 | file | yes, I gave the documentation above |
| 16:18.26 | monsterco | @file - thanks; let me explore that |
| 16:21.00 | consolejazz | file: hello. |
| 16:21.07 | consolejazz | file: does it make sense the approach I'm taking for install? |
| 16:21.15 | consolejazz | considering type of device I'm on and distro |
| 16:21.20 | consolejazz | join #voip |
| 16:21.29 | consolejazz | :/ |
| 16:21.33 | [TK]D-Fender | Yes |
| 16:21.36 | file | I don't have experience with that device, so I can't speak for how well it will work - but it should |
| 16:21.58 | consolejazz | [TK]D-Fender: howdy, was that in response to my general question? |
| 16:22.07 | [TK]D-Fender | consolejazz, YES. Go compuil as be the instructitons provided with * |
| 16:22.12 | [TK]D-Fender | compile* |
| 16:22.47 | consolejazz | [TK]D-Fender: perfect, thanks. Curious, on a typical modern desktop does the compile take a long while? |
| 16:22.56 | consolejazz | hasn't compiled any software in some time |
| 16:23.21 | [TK]D-Fender | not long |
| 16:23.48 | consolejazz | Also, I should be using SIP protocol for setup correct? |
| 16:23.59 | consolejazz | I'm subscribed to SIP trunk at voip.ms service |
| 16:24.55 | consolejazz | As for my ATA, Cisco SPA112, I'll have to sort that out in turn |
| 16:25.38 | [TK]D-Fender | yes |
| 16:25.45 | consolejazz | This is for small VoIP environment at home |
| 16:26.05 | consolejazz | couple softphones (Linphone on OS X) and then the ATA attached to cordless phone |
| 16:26.32 | consolejazz | forgive me If I'm butchering any VoIP terminology here; just starting out |
| 16:26.56 | [TK]D-Fender | not yet |
| 16:27.05 | consolejazz | :> |
| 16:32.41 | *** join/#asterisk warewolf (warewolf@warewolf.org) |
| 16:33.30 | warewolf | anybody know how turn on extended SSL debugging when troubleshooting SIP-TLS? |
| 16:34.10 | warewolf | I've got a hardware SIP phone working just fine w/ sip TLS, but linphone is complaining that there's an unsupported cipher. I can't tell what cipher linphone and asterisk are trying to use |
| 16:35.41 | *** join/#asterisk mirela666 (~mirkob@2a00:1950:400:0:c6e:f4f6:e502:b408) |
| 16:38.16 | warewolf | all I get on the asterisk side is |
| 16:38.18 | warewolf | tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 16:38.23 | warewolf | tcptls.c:684 handle_tcptls_connection: FILE * open failed! |
| 16:39.03 | monsterco | @file - I read somewhere (can't find now) to use my own Channel IDs wehn using ARI is preferred. Do you know why that is? and if I should use them are there any series of numbers I have to stay away from so I don't clash with other Asterisk calls? |
| 16:39.14 | monsterco | or this is totally different stuff and I don't have to worry about it at all |
| 16:39.23 | file | because you may receive events before your POST completes and returns |
| 16:39.43 | file | and you can use what you wish |
| 16:46.40 | *** join/#asterisk miralin (~Thunderbi@194.8.128.122) |
| 16:46.45 | monsterco | @file - makes total sense - thanks :) |
| 16:47.19 | *** join/#asterisk syadnom (~syadnom@167.88.114.244) |
| 16:47.33 | syadnom | hi all. I'm looking for a tool to monitor real-time latency on a sip channel. |
| 16:47.40 | syadnom | hopefully a simple CLI tool.... |
| 16:47.42 | monsterco | @file - on Asterisk CLI, other than setting verbose to 9; is there any other tool that helps me debug or see verbosity of ARI actions? |
| 16:47.56 | monsterco | syadnom - snmp? |
| 16:48.06 | syadnom | monsterco, I mean 'live'. |
| 16:48.13 | file | I'd join #asterisk-ari and ask there, so others can join in on how they figure out things |
| 16:48.20 | syadnom | as in , watch an in-flight call's latency |
| 16:48.57 | monsterco | listen via AMI maybe....not sure - but if this is to diagnose network I just check for patckets and quality (latency) of overall network and tell client their internet is bad... |
| 16:49.31 | monsterco | @file - didn't know that exists; thanks agian |
| 16:50.07 | monsterco | syadnom - oh....hmmm you mean once the channel is established? i think that still reflects on "sip show peer" - doesn't it? |
| 16:54.39 | syadnom | monsterco, yes, but then I've got logs scrolling through etc etc. |
| 16:58.22 | monsterco | Ooooo lol that is always the prob with CLI I find. Maybe you can use grep and | things at shell and do a script to check every second? |
| 16:58.47 | subvhome | are there any limitations with using AsteriskNOW? |
| 16:59.10 | subvhome | as apposed to building a server from scratch |
| 16:59.12 | subvhome | ? |
| 17:01.18 | nixnothing | "astersk"...NOW!? |
| 17:01.36 | file | AsteriskTHEN |
| 17:02.03 | nixnothing | you mean digium's paid support Asterisk solution? |
| 17:02.27 | subvhome | No.. there auto installer thing.. AsteriskNOW |
| 17:02.35 | nixnothing | ah |
| 17:02.36 | nixnothing | eww |
| 17:02.42 | file | AsteriskNOW is Asterisk + FreePBX |
| 17:03.08 | nixnothing | does it do weird config changes |
| 17:03.31 | [TK]D-Fender | GUI owns your configs, not you |
| 17:03.35 | nixnothing | or just install the packages? why would you not just use your pacman? |
| 17:03.47 | monsterco | AsteriskNOW is handled by Sangoma I think after they bought freepbx |
| 17:03.53 | nixnothing | GUI? |
| 17:03.54 | subvhome | you know .. im just going to go with no gui... |
| 17:04.15 | nixnothing | is this for winservers? |
| 17:04.20 | subvhome | thanks. |
| 17:04.23 | subvhome | no... |
| 17:04.26 | *** join/#asterisk rafaels (~rafaels@177.7.238.224) |
| 17:04.27 | nixnothing | yeah |
| 17:04.30 | nixnothing | the answer is |
| 17:04.34 | nixnothing | anything server demon |
| 17:04.38 | nixnothing | dont use GUI |
| 17:04.39 | nixnothing | ever |
| 17:04.42 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:04.46 | nixnothing | thats prob why I was confused |
| 17:04.54 | subvhome | well AsteriskNOW is a linux distro |
| 17:05.04 | nixnothing | eww |
| 17:05.06 | subvhome | so regardless of a gui being available.. the core is there |
| 17:05.19 | subvhome | I'm going with plain ol asterisk |
| 17:05.37 | subvhome | not a fan of the gewy |
| 17:05.53 | nixnothing | I don't like prebuilt specialized distros generally |
| 17:06.11 | *** join/#asterisk xnor (~alex@untian.silverninja.net) |
| 17:06.46 | nixnothing | *maybe I'm just biased tho* I run a cluster so gui would be automatic no |
| 17:07.04 | nixnothing | but I could see this for like a small office who wants to diy |
| 17:07.44 | nixnothing | sorry I didnt have context of the scope/environment you are using it foer |
| 17:08.02 | monsterco | nixnothing - u will trun around when you manage few more boxes :) |
| 17:08.07 | monsterco | or when user wants access... |
| 17:08.22 | nixnothing | ? |
| 17:08.28 | monsterco | GUI |
| 17:08.56 | nixnothing | we are centrailized cloud cluster so thats not a thing we ever have to worry about |
| 17:09.16 | nixnothing | we do have a lot of web monitorying tools tho |
| 17:09.41 | nixnothing | thier current software soltion is enswitch |
| 17:11.03 | nixnothing | its not the best overall solution if you were going to make a cloud voip cluster today, but thats what we use and have set up for like ~8 years. |
| 17:12.03 | nixnothing | basically the users all have a web portal to make changes to the thier own DB values so they can do it themselves if they want to |
| 17:12.45 | nixnothing | so I get what you mean, with GUI (im my case customer facing *graphical* web portal) |
| 17:13.25 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 17:14.06 | drmessano | It really depends on what youre doing |
| 17:14.19 | nixnothing | -_- they wont let me touch the perl code tho since enswitch wont support parts of It If I made my own custom changes to it tho.,.. sadness... T_T |
| 17:15.08 | drmessano | When someone says "I need a GUI to manage Asterisk", it's such an open ended request |
| 17:18.37 | nixnothing | heres how I would spit it up (small setups - GUI, graphical linux is fine) (medium / multi-tenant, below 1000 phones - some customization of asterisk, maybe grow to small cluster) (large, or cloud/centrailized cluster call bridging - heavier customization, code, solution, etc) |
| 17:18.51 | nixnothing | this is not considering call centers tho |
| 17:19.09 | nixnothing | thats a thing I wouldn't know much about |
| 17:20.32 | nixnothing | up to 100? small, single box, up to maybe 500/700 small cluster |
| 17:20.56 | nixnothing | *my guess |
| 17:21.57 | drmessano | Something like FreePBX is good if you are building a "PBX".. If you're using Asterisk as sort of a softswitch, then really what you need is a custom UI for changing those small things that need to be changed, like passwords |
| 17:22.23 | nixnothing | sure |
| 17:22.34 | nixnothing | I was thinking also like ehh what is it |
| 17:22.57 | drmessano | So the "I need a GUI" thing really revolves around your actual use of Asterisk |
| 17:23.02 | nixnothing | DAHDI vs something like rabbit for big clusters |
| 17:23.28 | *** join/#asterisk rafaels (~rafaels@177.7.238.224) |
| 17:23.32 | nixnothing | I mean whats "actual use" |
| 17:24.17 | nixnothing | I have never seen gui for any of this stuff |
| 17:24.27 | nixnothing | Its hard for me to visualize |
| 17:25.10 | nixnothing | does asteriskNOW have its own gui interface that is pretty plesant? |
| 17:25.53 | [TK]D-Fender | FreePBX <- |
| 17:25.54 | drmessano | Asterisk can be used for many things.. a traditional PBX, a "Voicemail server", maybe just the calling engine for some robodialer |
| 17:25.57 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 17:25.59 | [TK]D-Fender | And it is not multi-tennent |
| 17:26.22 | nixnothing | ? |
| 17:26.28 | [TK]D-Fender | nor something I'd consider " heavier customization" |
| 17:26.30 | nixnothing | multi-tennant is a business term |
| 17:26.36 | nixnothing | it has no crossover |
| 17:26.41 | nixnothing | its just jargon |
| 17:26.51 | [TK]D-Fender | It has a specific meaning in PBX world |
| 17:26.53 | drmessano | One of my favorite lines is "Asterisk is not a PBX, but can be used to build a PBX" |
| 17:27.19 | [TK]D-Fender | AsteriskNOW = CentOS + Asterisk + FreePBX |
| 17:27.26 | nixnothing | sure |
| 17:28.08 | nixnothing | I guess like same question |
| 17:28.55 | nixnothing | doe they just run as deamons (services) in the background, or do they have a plesent GUI interface on Graphical Linux |
| 17:29.09 | nixnothing | ive only used terminal |
| 17:29.45 | [TK]D-Fender | Nothing on the server itself |
| 17:29.54 | [TK]D-Fender | They tend to sit on a CLI loging you never use |
| 17:30.00 | nixnothing | so |
| 17:30.30 | nixnothing | AsteriskNOW is just for lazy ppl who dont know linux good? |
| 17:30.43 | drmessano | lol |
| 17:30.58 | [TK]D-Fender | How often do you admin any linux server at the physical keybaord & screen VS SSH? |
| 17:30.59 | nixnothing | cause you can just use whatever distro you like |
| 17:31.25 | drmessano | AsteriskNOW makes it easy to install those Asterisk + FreePBX |
| 17:31.26 | nixnothing | there is virtually no difference |
| 17:31.36 | [TK]D-Fender | AsteriskNOW is both for the linux-challenged as well as Asterisk challenged |
| 17:31.44 | drmessano | ^ |
| 17:31.50 | nixnothing | cool |
| 17:32.10 | nixnothing | makes sense |
| 17:32.13 | nixnothing | XD |
| 17:32.20 | drmessano | FreePBX is literally designed around configuring Asterisk as a PBX |
| 17:32.37 | drmessano | It's not WebMin or some crap that shortcuts a few config options |
| 17:32.52 | [TK]D-Fender | Software equivalent to the Toaster PBX's you can buy off-the-shelf |
| 17:33.10 | drmessano | It has its own dialplans that are written from GUI input.. |
| 17:33.23 | nixnothing | sure we use freePBX |
| 17:33.52 | drmessano | Then you know some people have a hard time installing FreePBX |
| 17:33.57 | drmessano | and even Asterisk |
| 17:34.15 | nixnothing | I forget sometimes you don't have to use asterisk(asterisk setups) with only a few features |
| 17:34.25 | drmessano | Even with the FreePBX wiki now, people still don't follow the many guides STEP by STEP |
| 17:34.49 | nixnothing | no |
| 17:34.52 | drmessano | So AsteriskNOW, and more preferebly the FreePBX Distro make it easy |
| 17:34.54 | nixnothing | I use it |
| 17:35.02 | nixnothing | but I didnt set the cluster u[ |
| 17:35.04 | nixnothing | up |
| 17:35.34 | nixnothing | I had to do a bunch of research when I first started to work backwards |
| 17:36.03 | nixnothing | I had used Asterisk in a limited context |
| 17:36.14 | nixnothing | not as viable PBX |
| 17:36.34 | [TK]D-Fender | I use Asterisk as jukebox and to make me coffee |
| 17:36.34 | nixnothing | but just messing around with it and setting up a some dialplans |
| 17:36.36 | nixnothing | and stuff |
| 17:36.56 | nixnothing | tying in to g-voice, softphone sip clients |
| 17:39.21 | drmessano | g-voice is kind of a waste at this point |
| 17:40.30 | nixnothing | eh, I like gvoice, its easy to use and gives you free #s |
| 17:46.40 | drmessano | Undocumented API that has every chance of being turned off at a moments notice |
| 17:46.47 | drmessano | Useless investment |
| 17:46.59 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP140-03-1177760945.dsl.bell.ca) |
| 17:47.01 | drmessano | Google is divesting itself of XMPP.. |
| 17:47.25 | nixnothing | its not an investment |
| 17:47.48 | nixnothing | its like click this button to get texts on your browser and all your devices |
| 17:47.54 | nixnothing | *click* |
| 17:48.21 | drmessano | Sure it is.. You're using it.. you're vested. If it gets turned off tomorrow, you're have to waste time moving to something else |
| 17:48.23 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 17:48.35 | nixnothing | not really |
| 17:48.43 | nixnothing | they give you that number |
| 17:48.55 | nixnothing | you can x-fer it to a carrier at any time |
| 17:49.04 | drmessano | Well aware |
| 17:49.31 | nixnothing | and crome has this cool plugin so I can text people from my browser at work |
| 17:49.36 | drmessano | So if Google turns off XMPP tomorrow, porting takes 10 seconds? |
| 17:49.42 | nixnothing | its all personal preference man |
| 17:49.59 | nixnothing | dude |
| 17:50.02 | drmessano | Google turning off an API is not personal preference |
| 17:50.02 | nixnothing | dont get salty |
| 17:50.13 | drmessano | Nobody is getting salty |
| 17:50.27 | nixnothing | personal preference |
| 17:50.32 | drmessano | Google turning off an API is not personal preference |
| 17:50.36 | nixnothing | you can have yours |
| 17:51.03 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-13.wall.uls.co.za) |
| 17:51.10 | drmessano | Ok, well.. Youre not even making sense now.. so whatever |
| 17:51.16 | nixnothing | lol |
| 17:52.27 | nixnothing | sorry if I offended you man. Nothing was a personal jab |
| 17:52.49 | drmessano | Didnt offend me |
| 17:52.55 | drmessano | Didnt make me salty |
| 17:53.12 | nixnothing | ... |
| 17:53.15 | drmessano | I was stating a fact, and you were making it an opinion |
| 17:53.16 | nixnothing | im trying to be nice |
| 17:53.24 | [TK]D-Fender | continues seasoning drmessano and fires up the grill |
| 17:53.29 | drmessano | lol |
| 17:53.33 | nixnothing | you were stating you oppinion |
| 17:53.37 | drmessano | No I wasnt |
| 17:53.42 | [TK]D-Fender | Jeff's nuts roasting on an open fire.... |
| 17:53.44 | nixnothing | I never stated a fact to refute |
| 17:53.54 | [TK]D-Fender | carols |
| 17:54.04 | nixnothing | I just said "I use gvoice and I like it" |
| 17:54.06 | drmessano | Fact: Google Voice in Asterisk uses XMPP |
| 17:54.16 | drmessano | Fact: Google is turning off XMPP |
| 17:54.17 | nixnothing | there is nothing to prove me wrong on |
| 17:54.24 | nixnothing | I agree |
| 17:54.32 | drmessano | You dont need to agree |
| 17:54.33 | nixnothing | why are you agueing |
| 17:54.34 | drmessano | those are facts |
| 17:54.39 | nixnothing | sure |
| 17:54.49 | nixnothing | they dont facter in to anything I said tho |
| 17:55.01 | [TK]D-Fender | Fact: Chocolate is better than vanilla. If you disagree then you are simply WRONG. |
| 17:55.03 | nixnothing | I said "I like this" |
| 17:55.51 | nixnothing | you said "I don't" |
| 17:55.53 | nixnothing | thats cool |
| 17:55.55 | drmessano | Actually you indicated you had spent time "tying into g-voice" as you put it |
| 17:55.58 | [TK]D-Fender | nixnothing, Like it for now and don't be surprised when the free ride disappears. If you'good with that then carry on... |
| 17:56.04 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
| 17:56.08 | drmessano | Actually I never said I didnt like it |
| 17:56.13 | drmessano | Now you are putting words into my mouth |
| 17:56.13 | nixnothing | yea.... for fun |
| 17:56.17 | nixnothing | not to actually use |
| 17:56.21 | nixnothing | that would be silly |
| 17:56.32 | nixnothing | when I was first tinkering w/ asterisk |
| 17:57.00 | nixnothing | this conversation is silly |
| 17:57.08 | nixnothing | dude |
| 17:57.10 | drmessano | This is making my head hurt, so I am going to go do something productive.. like argue which Bear is best. |
| 17:57.11 | nixnothing | I hate mondays |
| 17:57.18 | nixnothing | how about them nets? |
| 17:57.21 | malcolmd | Yogi |
| 17:57.58 | [TK]D-Fender | drmessano, Care clearly. |
| 17:57.58 | nixnothing | ITs OBVIOSLY smokey |
| 17:58.05 | drmessano | FALSE, Black bear |
| 17:58.12 | nixnothing | anly you can prevent forest fires |
| 17:58.33 | drmessano | FACTS: Bears eat Beets |
| 17:58.37 | drmessano | FACT* |
| 17:58.49 | [TK]D-Fender | can barely function with the guilt of all the forest fires he has failed to prevent... |
| 17:58.57 | drmessano | Bears, beets, battlestar galactica |
| 17:59.08 | [TK]D-Fender | Connect. The. Dots. |
| 17:59.20 | [TK]D-Fender | looks for his tin-foil hat |
| 18:00.05 | drmessano | Identity theft is not a joke, Jim |
| 18:00.12 | nixnothing | foil causes microwave fires D-F |
| 18:00.26 | drmessano | Not true |
| 18:00.26 | nixnothing | and if the microwave is near a forest......well |
| 18:00.40 | drmessano | A Kylstron causes Microwave fires |
| 18:00.45 | drmessano | Klystron |
| 18:00.48 | [TK]D-Fender | drmessano, I told you ... you're going on the grill... |
| 18:01.40 | nixnothing | over my dead rib's |
| 18:01.41 | drmessano | or a Magnetron |
| 18:01.54 | drmessano | Depends on the application |
| 18:01.58 | nixnothing | I only room for ribs on the grill |
| 18:02.37 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 18:02.47 | drmessano | But the Klystron or Magnetron is responsible for the fire.. The foil only provides the fuel |
| 18:03.20 | [TK]D-Fender | Ribs : meant to protect the animal in question yet is the very reason we kill several animals. #morissette |
| 18:03.47 | drmessano | [TK]D-Fender: Isn't it Ironic? |
| 18:03.54 | drmessano | Except, not really "Irony" |
| 18:04.01 | [TK]D-Fender | Don't you think? |
| 18:04.19 | drmessano | Ironic that a song about Irony is ironically not at all about Irony |
| 18:04.49 | [TK]D-Fender | Irony : When your cloths are a little more crinkled than they should be... |
| 18:05.21 | drmessano | She knows nothing about Irony but she sure can fold! |
| 18:05.25 | drmessano | Ba-dump-ching |
| 18:05.47 | nixnothing | maybe thats why they taste so delicious |
| 18:05.56 | nixnothing | irony is the best spice |
| 18:06.24 | nixnothing | oh |
| 18:06.25 | nixnothing | ...... |
| 18:06.28 | nixnothing | ironing |
| 18:06.33 | nixnothing | .....*facepalm* |
| 18:06.42 | subvhome | I wonder how many phones can work on RasPBX... as far as reaching hardware limitations. |
| 18:07.50 | drmessano | Phones are not a metric |
| 18:07.57 | nixnothing | concurrent calls |
| 18:08.09 | nixnothing | or incomming |
| 18:08.27 | drmessano | I would do many 15 concurrent calls on a Pi, no recording |
| 18:08.32 | drmessano | maybe* |
| 18:08.36 | *** part/#asterisk aisrael (~aisrael@pylon.battleaxe.net) |
| 18:08.37 | drmessano | A Pi 3 |
| 18:08.46 | subvhome | interesting. |
| 18:09.16 | subvhome | how is this measured... like how do i know what kind of server i would need... My budget is about 5k |
| 18:09.36 | drmessano | How many endpoints, concurrent calls? |
| 18:09.46 | drmessano | Recording? COnferences, Voicemail? |
| 18:10.19 | subvhome | drmessano.. is there are documentation i can reference when i get these figures? |
| 18:10.28 | nixnothing | Phones? Users? Expected max of ppl calling in at the same times? Queues or IVRs? |
| 18:10.48 | drmessano | subvhome: No |
| 18:10.58 | drmessano | It's having know-how |
| 18:11.04 | nixnothing | ^ |
| 18:11.23 | drmessano | You dont have ANY of those figures? |
| 18:11.28 | subvhome | ok.. good to know what factors in.. I can't wait to get started.. Getting everything i need first before i dive into this. |
| 18:11.32 | nixnothing | like you could find it, but you would understand how to use that info |
| 18:12.08 | nixnothing | =D |
| 18:12.20 | nixnothing | its fun stuff |
| 18:12.32 | drmessano | You also need to know how those calls are going to get to the PBX. PRI? SIP? Analog-o-Fun? |
| 18:12.46 | subvhome | 12 phones.. 12 concurrent calls... |
| 18:13.00 | subvhome | vm recording. |
| 18:13.16 | drmessano | You dont need much then |
| 18:13.38 | drmessano | Something like a SuperMicro server with a pair of drives |
| 18:13.53 | drmessano | Drop 4GB of RAM into it, because its cheap |
| 18:14.07 | [TK]D-Fender | drmessano> Phones are not a metric <- everyone knows they are IMPERIAL |
| 18:14.27 | nixnothing | ooooohhhh |
| 18:15.07 | nixnothing | *facepalm* |
| 18:15.47 | drmessano | [TK]D-Fender: I bought 2 litres of Water and it cost 4 pounds, I was so taken back I dropped it on my feet and fell back a meter. |
| 18:16.13 | [TK]D-Fender | Should have paid in euro to match... |
| 18:16.36 | drmessano | [TK]D-Fender: Nobody likes using the Euro, except the banks |
| 18:16.38 | nixnothing | did you guys hear about that TCP joke? |
| 18:16.46 | [TK]D-Fender | Brit currency = imperial, not metric |
| 18:16.46 | drmessano | GAH |
| 18:16.47 | drmessano | NO |
| 18:16.55 | [TK]D-Fender | nixnothing, ACK |
| 18:17.28 | drmessano | It's "Someone told me a UDP joke, but I didn't get it, and I wasn't worried about it" |
| 18:17.46 | nixnothing | ah I was gonna say |
| 18:17.54 | nixnothing | "you'll get it eventually" |
| 18:18.15 | drmessano | That would be a TCP joke |
| 18:18.21 | nixnothing | yeah |
| 18:18.45 | drmessano | You messed up a Layer 3 joke |
| 18:18.59 | nixnothing | 14:16 < nixnothing> did you guys hear about that TCP joke? |
| 18:19.13 | drmessano | (Except that wasnt Layer 3) |
| 18:20.12 | [TK]D-Fender | Joke delivery = Layer 1 problem |
| 18:20.30 | nixnothing | it was just a joke |
| 18:21.56 | [TK]D-Fender | And we are carrying it to its natural conclusion and a glorious funeral pyre |
| 18:22.01 | drmessano | I had a great Asterisk 1.2 joke once. I wrote it down on a piece of paper, and it just fell to the floor |
| 18:22.07 | drmessano | It was no longer supported |
| 18:22.09 | nixnothing | also |
| 18:22.24 | nixnothing | layer doesnt really matter |
| 18:22.41 | drmessano | Layer always matters |
| 18:23.01 | nixnothing | only when your sending something over a network |
| 18:23.15 | drmessano | Isnt that what we're doing, RIGHT NOW? |
| 18:23.24 | *** join/#asterisk coppice (~chatzilla@123.203.240.102) |
| 18:23.24 | nixnothing | also they are both 4 |
| 18:23.25 | [TK]D-Fender | Dean Kamen installed Asterisk 1.2 on the last model he rode. It Seg Faulted. |
| 18:23.31 | nixnothing | which is transport |
| 18:23.39 | nixnothing | you are still wrong |
| 18:24.01 | nixnothing | I just dint want to argue bucause jokes are supposed to be fun and happy |
| 18:24.10 | nixnothing | and haha |
| 18:24.14 | drmessano | Who is still wrong? |
| 18:24.23 | nixnothing | they are both layer 4 |
| 18:24.45 | drmessano | Ok, nobody said they werent |
| 18:24.59 | malcolmd | I had a great Asterisk 1.2 joke once. I wrote it down on a piece of paper| and it just fell to the floor |
| 18:25.12 | malcolmd | ;) |
| 18:25.23 | drmessano | It was no longer supported! |
| 18:25.43 | nixnothing | 14:18 < drmessano> That would be a TCP joke |
| 18:25.44 | nixnothing | 14:18 < nixnothing> yeah |
| 18:25.44 | nixnothing | 14:18 < drmessano> You messed up a Layer 3 joke |
| 18:25.44 | nixnothing | 14:18 < nixnothing> 14:16 < nixnothing> did you guys hear about that TCP joke? |
| 18:25.44 | nixnothing | 14:19 < drmessano> (Except that wasnt Layer 3) |
| 18:25.54 | drmessano | I know what I said |
| 18:26.03 | drmessano | WASNT |
| 18:26.17 | drmessano | I was trying to see if you would catch that |
| 18:26.21 | nixnothing | so you cant mess up a joke |
| 18:26.22 | nixnothing | also |
| 18:26.25 | nixnothing | your a dick |
| 18:26.27 | drmessano | Clearly it took a google search |
| 18:26.43 | drmessano | "you're" |
| 18:26.44 | malcolmd | i'm just pointing out asterisk 1.2 and pipes |
| 18:26.54 | malcolmd | because they're not commas |
| 18:27.02 | malcolmd | and because it's hilarious |
| 18:27.14 | drmessano | malcolmd: HA.. Yes, I missed that one. and wow, the bad memories |
| 18:27.23 | nixnothing | I just didnt want to embaress you |
| 18:27.24 | malcolmd | >:) |
| 18:27.48 | drmessano | nixnothing: Nothing to embaress, I know my OSI layers |
| 18:27.50 | drmessano | As should you |
| 18:27.58 | nixnothing | b/c there doesn not need to be conflic |
| 18:28.20 | drmessano | Doesnt not need = Does need |
| 18:28.28 | drmessano | THats a double-negative |
| 18:28.32 | nixnothing | learn to speel |
| 18:28.33 | drmessano | Why you frontin? |
| 18:29.17 | drmessano | 14:26:25 <nixnothing> your a dick |
| 18:29.26 | drmessano | You're retreaded |
| 18:29.40 | [TK]D-Fender | Comma : a brief pause. Coma : a very long pause. |
| 18:29.48 | *** join/#asterisk rafaels (~rafaels@177.7.238.224) |
| 18:31.12 | *** join/#asterisk neonerz (~G_Bove@cpe-24-88-251-67.nc.res.rr.com) |
| 18:31.45 | drmessano | F**kin A dog; F**kin A, dog.. Punctuation can keep you out of prison |
| 18:32.29 | neonerz | running asterisk 11, can anyone point me to a way to convert a mp4 into a format asterisk can read for video? It's encoding in h264, but I can't figure out how to get asterisk to play it. If I record the video directly in asterisk it saves it as .h263 or .h264 which then I could easily playback |
| 18:32.58 | neonerz | I basically have a video I made that I want to be able to play on one of my phones. |
| 18:33.36 | [TK]D-Fender | "Help your Uncle Jack off his horse". "Help your Uncle jack off his horse". Wht a difference a capital can make... |
| 18:34.13 | *** join/#asterisk areski (~areski@80.174.128.47.dyn.user.ono.com) |
| 18:34.18 | drmessano | [TK]D-Fender: ROFL |
| 18:35.35 | [TK]D-Fender | is the de-facto winner of every game of Cards Against Humanity he participates in, and at least 3 he hasn't. |
| 18:37.35 | nixnothing | love cards against humanity |
| 18:38.14 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 18:39.23 | nixnothing | theres a way to play online |
| 18:40.43 | nixnothing | that comment 14:31 was pretty ... distastful |
| 18:40.52 | nixnothing | wonder if dr is a redneck?.. |
| 18:41.09 | *** join/#asterisk KerioMorgan (~Adium@us.kerio.com) |
| 18:48.30 | [TK]D-Fender | It's only a minor sunburn and possibly some ketchup |
| 18:48.47 | nixnothing | aparently freeswitch is supposed to be better than asterisk at cloud based solutions (+ concurrent calls), but asterisk is still the goto for small setups |
| 18:48.56 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 18:49.01 | nixnothing | wan't to try stress tests on both down the line |
| 18:49.33 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 18:51.47 | *** join/#asterisk obelixBE (~obelix@2a02:1811:c52c:1700:4d8a:834e:c039:97d0) |
| 18:52.14 | nixnothing | ..so bored... maybe I should go do some programming. I've been thinking of doing project that puts a twist on conway's game of life. |
| 18:53.56 | nixnothing | I'll be back to guide the poor souls of randoms who wonder in later. |
| 19:02.18 | neonerz | does anyone know of any tools that would allow me to convert a video into a format asterisk likes natively? (running asterisk 11) |
| 19:03.38 | nixnothing | videos are intersting on asterisk |
| 19:03.41 | neonerz | it doesn't seem to like an mp4 without installing a third party app like app_mp4 |
| 19:03.50 | nixnothing | but Im sure there are some codec conversion tools |
| 19:04.12 | neonerz | and I can't figure out how to convert it to a file like .h264 or .h263 which asterisk seems to support natively |
| 19:06.28 | nixnothing | do some research on codec sonversion tools on linux. At worst you may have to write a small script depending on what your are doing. If you are trying to send an already saved video, it will be easier. But for something like video chats, your are goind to want to find a way to record in that format |
| 19:07.00 | neonerz | it's not something that needs to be done on the fly, I just want to convert already made video to use within a IVR or some other application. |
| 19:07.30 | neonerz | I assume maybe ffmpeg, but I can't figure out how to output as a raw .h264 or whatever file without using a container like .mp4 or .mk |
| 19:07.45 | nixnothing | yeah just look for some linux video/codec conversion tools |
| 19:07.45 | *** join/#asterisk sh1rtybird (~quassel@beepbeep.serverpit.com) |
| 19:08.01 | neonerz | And recording it directly from the phone isn't an option. |
| 19:08.25 | nixnothing | you can prob configure your setup so that you can just dump videos in one directory |
| 19:08.42 | nixnothing | and then have a tool to check & convert them |
| 19:09.36 | nixnothing | just make sure that when they are converted to plan the use the converted file in the IVR |
| 19:11.09 | neonerz | that's the issue, I can't find a tool to convert them in a format Asterisk likes. |
| 19:14.21 | *** join/#asterisk CheBuzz (~CheBuzz@unaffiliated/chebuzz) |
| 19:21.02 | *** join/#asterisk defsdoor (~andy@cpc19-sutt4-2-0-cust102.19-1.cable.virginm.net) |
| 19:26.42 | drmessano | Your problem is your google foo |
| 19:27.26 | drmessano | Youre looking based on codec |
| 19:27.40 | drmessano | Most tools are based on file format. |
| 19:29.50 | *** join/#asterisk K0HAX (~michael@2001:470:c298:1::20) |
| 19:33.20 | neonerz | drmessano: I don't disagree with that, do you have a better direction I should be looking? |
| 19:35.02 | neonerz | correct me if I'm wrong, but isn't a .h264 just raw video in the h.264 codec? So in this situation file format and codec are the same thing? |
| 19:36.29 | drmessano | neonerz: I'm looking, actually.. Im more familiar with audio types.. I did find this though |
| 19:36.36 | drmessano | The h263 files that Asterisk plays are not portable from other systems. |
| 19:36.36 | drmessano | They are merely a collection of frames sent by Asterisk, and saved into |
| 19:36.36 | drmessano | a file. The use case for this is to record video voicemail and play the |
| 19:36.36 | drmessano | video voicemail back. You cannot take H.263 files from other systems |
| 19:36.36 | drmessano | and expect them to play correctly on Asterisk. |
| 19:36.47 | drmessano | Whoa.. sorry |
| 19:37.33 | neonerz | so then is something like the third party app app_mp4 my only way of doing this? I can't be the first person to ever want to create a video IVR from video not recorded directly in Asterisk. |
| 19:37.53 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 19:38.59 | neonerz | (this is app_mp4 I'm talking about if you aren't familiar: http://www.medooze.com/products/asterisk-applications/app_mp4.aspx) |
| 19:39.14 | drmessano | Yeah.. I just found that |
| 19:39.57 | neonerz | voip-info seems to indicate that asterisk has native support for mpeg4 since asterisk 10, but we all know how reliable that site can be :) |
| 19:40.05 | neonerz | http://www.voip-info.org/wiki/view/Asterisk+video |
| 19:40.37 | drmessano | Im sure it does |
| 19:40.55 | drmessano | But we're not talking about bridging channels |
| 19:41.31 | drmessano | Someone wrote a little patch to make vp8 work, but that's for negotiating calls between two endpoints |
| 19:41.44 | drmessano | What you're talking about is really very different.. and |
| 19:41.53 | drmessano | You may need something external |
| 19:42.23 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 19:44.02 | *** join/#asterisk rafaels (~rafaels@177.82.136.167) |
| 19:44.50 | drmessano | I like what you're trying to do here |
| 19:45.00 | drmessano | Im curious myself as to how to make this work |
| 19:45.01 | *** join/#asterisk K0HAX (~michael@shellhost.home.englehorn.com) |
| 19:45.12 | drmessano | Im just not seeing the glue |
| 19:46.58 | neonerz | yea, same here. But I feel like it has to be possible. There seemed to be some old app called ffasterisk that would convert video for asterisk, but the company that made it seemed to remove if from their site |
| 20:02.05 | xnor | have any of ya'll worked with the app_jack module? I'm getting a lot of NaN on the output.. wondering where to start debugging that |
| 20:08.12 | drmessano | Havent thought of app_jack in years |
| 20:08.18 | drmessano | Does it still work? |
| 20:08.23 | *** join/#asterisk mjordan (mjordan@nat/digium/x-ckgltdvfxtukqrnd) |
| 20:08.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:08.52 | drmessano | mjordan: Glad to see you finally decided to learn VoIP |
| 20:08.55 | xnor | drmessano: i am able to route calls with it but the data is non-deterministically messed up |
| 20:08.59 | drmessano | (LinkedIn) |
| 20:11.44 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:12.40 | nixnothing | not to familiar |
| 20:13.06 | nixnothing | can research and help troubleshoot if you give me a couple mins |
| 20:14.29 | nixnothing | how are you ising app_jack? |
| 20:15.00 | nixnothing | ive seen something similar |
| 20:15.29 | nixnothing | are you using it for input or output |
| 20:16.08 | mjordan | drmessano: I figured I should bother at some point. |
| 20:16.13 | mjordan | suspects LinkedIn |
| 20:16.59 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 20:17.03 | xnor | nixnothing: i'm using it for both, i have asterisk creating a jack input/output then using jack_connnect to connect to pure data which is where i'm detecting the nan |
| 20:17.25 | drmessano | mjordan: Well, maybe after you've proven some sort of expertise in the area I can endorse you for that skill. |
| 20:17.27 | xnor | i'm running jackd -R -d dummy |
| 20:17.39 | xnor | oops, jackd -R -d dummy -r 44100 |
| 20:17.46 | xnor | i should probably lower the sample rate.. .. |
| 20:18.28 | xnor | nixnothing: unfortuantely i'm at work right now so i'm not in front of my setup.. |
| 20:18.44 | nixnothing | ditto |
| 20:19.28 | nixnothing | if its non-ugent I can work on it on my end |
| 20:19.48 | xnor | nixnothing: cool, thanks a ton! |
| 20:20.13 | nixnothing | I usually idle on here all the time and just reconnect to a screen session when im at work |
| 20:21.09 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 20:21.12 | xnor | thats bascially my approach as well, though i get back on when i'm working on after work projects as well, which is what this is [http://futel.net] |
| 20:21.30 | xnor | i'm gonna get back to debugging it later this evening [~6pm PST] |
| 20:21.50 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 20:21.54 | nixnothing | lol |
| 20:22.01 | nixnothing | this reminds me of a friends project |
| 20:22.31 | xnor | links? |
| 20:23.08 | nixnothing | he made a video library web interface that a bunch of videos to watch between friends |
| 20:23.23 | nixnothing | he endedup getting the domain |
| 20:23.27 | nixnothing | quicktime.club |
| 20:23.32 | nixnothing | just as a troll name |
| 20:23.45 | Micc | Is it a known vulnerability with chan_sip that it would allow an invite from an IP address that was not registered when insecure=invite? From what I can tell inscure=invite just allows the invite if that IP had already authenticated with a register. But I have in asterisk 1.8.9.2 an example of a completely separate IP making a call. |
| 20:23.46 | xnor | nice |
| 20:23.52 | nixnothing | like "dude, don't you just LOVE quicktime!" |
| 20:24.25 | drmessano | xnor: I just have one question about your project |
| 20:24.29 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 20:24.33 | xnor | drmessano: yes? |
| 20:25.05 | drmessano | xnor: When you re-deploy a pay phone as a Futel phone, is the urine smell simulated or do you actually re-christen the endpoints? |
| 20:25.58 | xnor | you know, we should re-christen :) |
| 20:25.58 | Micc | I could not find a corresponding vulnerability listed in the asterisk security advisories. |
| 20:27.09 | drmessano | Micc: Sounds to me like insecure is doing exactly what it's supposed to do |
| 20:28.02 | drmessano | Micc: Are you setting insecure on an endpoint? |
| 20:28.22 | nixnothing | Might have been a false positive |
| 20:28.34 | nixnothing | do you have an example |
| 20:29.21 | nixnothing | *the |
| 20:29.30 | Micc | drmessano, setting on the sip peer in sip.conf. |
| 20:29.55 | *** join/#asterisk mirela666 (~mirkob@52D9ADEB.cm-11-1c.dynamic.ziggo.nl) |
| 20:30.04 | drmessano | Micc: Ok, and you should be dumping those calls into a context that can't route back out and cost money |
| 20:31.03 | drmessano | Your provider is not going to authenticate to you, so all inbound calls should be treated as hostile, basically |
| 20:31.10 | Micc | drmessano, I have some customers with phones that require insecure=invite, but I would expect it to check the IP the invite comes from at least. |
| 20:31.43 | drmessano | Micc: What scenario requires that?? |
| 20:32.50 | drmessano | That is basically a "Bend over, here's the lube" kinda nightmare scenario |
| 20:32.50 | Micc | drmessano, certain phones behind certain firewalls with certain firmwares. |
| 20:33.05 | Micc | why wouldn't asterisk just check the IP? |
| 20:33.51 | Micc | I have never read anywhere that insecure=invite has this kind of vulnerability. |
| 20:34.02 | *** join/#asterisk chazzam (~chazz@vps.roguewerks.com) |
| 20:36.35 | scv | it's 'insecure=invite' for a reason |
| 20:45.06 | Micc | the documentation for insecure=invite doesn't say that though. |
| 20:45.17 | Micc | It just says it won't require authentication |
| 20:45.30 | Micc | that is different. |
| 20:46.35 | Micc | and insecure=port is for matching by ip address without matching port number. |
| 20:46.59 | Micc | So I would assume that insecure=invite still matches by ip address and port, but just doesn't require authentication. |
| 20:48.40 | *** join/#asterisk jazper- (~kcd@pdpc/supporter/active/jazper) |
| 20:49.24 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
| 20:51.38 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:52.22 | *** join/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
| 20:52.28 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 20:57.11 | wdoekes | Micc: as far as I know insecure=invite indeed checks the ip+port and then skips the auth |
| 20:57.55 | wdoekes | are you sure the call wasn't authed when it was accepted from the other ip? |
| 20:59.02 | Micc | yes. sip show peer on it shows a different ip. I'll copy and paste the sip debug and cli. |
| 21:01.06 | wdoekes | you didn't answer the question |
| 21:01.54 | wdoekes | sip show peer shows from where it was registered, not whether the invite contained authentication or not |
| 21:05.30 | Micc | I guess I don't know the answer to that except that I also saw authentications failed from this same ip as well. http://pastebin.com/NNcxK8dX |
| 21:14.27 | wdoekes | type=friend or type=peer? |
| 21:15.51 | *** join/#asterisk KerioMorgan (~Adium@us.kerio.com) |
| 21:17.00 | Micc | type=friend |
| 21:19.27 | wdoekes | a quick glance at the source hints that user-matching will indeed bypass the ip-port check if you have insecure=invite |
| 21:19.49 | wdoekes | cannot tell for sure, but you make a compelling case |
| 21:21.29 | Micc | If I had read anywhere that it would bypass the ip check, I would have never used it. |
| 21:21.42 | Micc | let me see what asterisk book says. |
| 21:22.45 | wdoekes | for your case, type=peer would've been right |
| 21:22.52 | wdoekes | *appropriate |
| 21:25.27 | wdoekes | looks to me like it got type=user insecure=invite got unintentionally added here: |
| 21:25.30 | wdoekes | commit c761bea8e2f3d2c8dca31ddf100dc0f6eea008f8 |
| 21:25.32 | Micc | I think I changed to type=peer at some point, but I read somewhere it was almost the same so I never went back and changed it in older configs. |
| 21:25.33 | wdoekes | Author: Luigi Rizzo <rizzo@icir.org> |
| 21:25.35 | wdoekes | Date: Mon Oct 23 11:08:47 2006 +0000 |
| 21:27.06 | wdoekes | well, it's not |
| 21:27.06 | drmessano | wdoekes: That might be a really good 10 year old issue to sew up in a grand fashion lol |
| 21:28.18 | wdoekes | the type=user matches the From-user-part, the type=peer matches the ip-port when selecting which asterisk-user/peer it is who is doing the INVITE |
| 21:28.49 | drmessano | creating a user is the problem, and the somewhat "accidental" creation of the user with type=friend. Thats why I generally WARN people about using type=friend without knowing the risks |
| 21:29.07 | drmessano | "Accidental" as in "Not knowing what 'friend' does" |
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| 21:29.33 | Micc | The note I see in the book is that type=friend will match on username first, IP second. |
| 21:29.37 | Micc | that sounds fine to me. |
| 21:30.15 | wdoekes | and that insecure=invite will disable any auth after doing the matching? |
| 21:30.26 | drmessano | Correct |
| 21:30.31 | mjordan | From sip.conf: ';insecure=invite ; Do not require authentication of incoming INVITEs' |
| 21:30.55 | mjordan | goes back to PJSIP land |
| 21:31.16 | wdoekes | hehe |
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| 21:31.35 | drmessano | So with friend you have a user getting matched FIRST, with no auth, and no IP match because the user is there |
| 21:31.53 | drmessano | Open door, come right in, beer in the fridge |
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| 21:35.41 | Micc | drmessano, I guess I assumed it would always match the IP secondly, only after matching the user, then it would also match the ip. |
| 21:37.00 | Micc | I see how that makes sense now, but it wasn't clear to me that matching on IP second would only happen if matching of the user failed. |
| 21:37.03 | drmessano | Well now you know |
| 21:37.26 | Micc | which I would also think is bad. I want to match on both always. |
| 21:37.44 | drmessano | type=friend became the shoehorn of configuring a SIP device. "It's gonna work one way or another, but dammit it's going to work!" |
| 21:38.34 | drmessano | Which is fine, I GUESS MAYBE, if your dialplan looks like a maze with no end that results in a context that can put the call back out to another peer. |
| 21:38.56 | Micc | It says friend matches both. |
| 21:39.02 | drmessano | No |
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| 21:39.09 | drmessano | friend creates both |
| 21:39.37 | drmessano | and matches either, in order of username then ip |
| 21:39.39 | Micc | "This enables matching rules for both peer and user. This is the setting most commonly used for SIP phones." |
| 21:39.39 | wdoekes | Micc: you may want to match AND-style on both, but many devices send a CLI in the From-part, so matching on that would fail |
| 21:40.21 | drmessano | Micc: "both" is more "or" |
| 21:41.26 | Micc | also, page 104 asterisk book: "The first option we've configured is the type, which we've set to friend. This tells the channel driver to attempt to match on the name first, AND then IP address." |
| 21:42.00 | Micc | so you see how I could be confused here. |
| 21:42.09 | wdoekes | "attempt to match" does not sound like strict checking to me |
| 21:42.26 | drmessano | But matching is exclusive |
| 21:43.13 | drmessano | attempt to match name, then attempt to match IP.. implying failure if the first isnt satisfied |
| 21:43.33 | drmessano | Poorly worded, no doubt |
| 21:44.37 | Micc | Even if it was worded correctly, I still would have been confused because nothing I've read on inscure=invite metioned this possibility. |
| 21:44.56 | wdoekes | in any case, for future poor souls attempting this, docs could certainly get an extra warning here and there. but reverting the 10y/o behaviour doesn't sound like a viable option |
| 21:45.25 | drmessano | wdoekes: reverting the 10y/o behaviour doesn't sound like a viable option <--- NO FEAR |
| 21:45.51 | Micc | I'm fine with the way that it is if I know how it works, like I do now. |
| 21:46.01 | drmessano | or add "insecure=mostly" |
| 21:48.06 | Micc | Something about this should be mentioned in README-SERIOUSLY.bestpractices.txt |
| 21:48.15 | Micc | as well as update the sip.conf comment and the asterisk book. |
| 21:48.45 | wdoekes | please file an issue on the bug tracker and add the textual patches you would add to sip.conf.sample and README-SERIOUSLY.bestpractices.txt |
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| 22:32.04 | syadnom | guys, any solution to: Received SIP subscribe for peer without mailbox? |
| 22:32.08 | syadnom | filling my logs? |
| 22:32.12 | syadnom | other than externotify? |
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| 22:40.18 | [TK]D-Fender | Add the mailbox? |
| 22:40.24 | [TK]D-Fender | Stop the client? |
| 22:40.28 | [TK]D-Fender | Filter your logs? |
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