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| 02:34.56 | smd75jr_ | I'm back, and after quite a bit of poking and prodding, I can now dial out AND receive inbound calls, however, the reason it works appears to be due to an ugly hack of a dialplan that looks like this: (xx.<@10.0.1.1>) (where 10.0.1.1 is the asterisk server). At the moment the system is forcing me to enter an extension number even though I have a ring group set up that should automatically route anything on this DID to extension 0001. |
| 02:34.57 | smd75jr_ | <PROTECTED> |
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| 03:25.55 | ChannelZ | Having not been here all day I'd need more to go on than that |
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| 03:32.13 | smd75jr_ | Haha, sorry, what basically happened was that the spa3000 was refusing to route incoming calls through to the asterisk server (coming from the pstn line), I finally was able to get it to sort of work using the above dialplan |
| 03:33.15 | smd75jr_ | but now it makes me dial an extension number when it connects, despite the face that I have a ring group set up to send that particular trunk directly to extension 0001 |
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| 03:57.54 | r00f | not familiar with this particular device, but usually you set extension number via hotline, then put in the route (digit map) that this extension goes to sip, and in incoming destination you choose hotline, not extension |
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| 03:58.39 | r00f | i remember i was not able to set up without support, their configuration logic is non-human |
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| 03:58.49 | smd75jr_ | hotline? |
| 03:58.57 | smd75jr_ | as in S0 |
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| 04:06.01 | smd75jr_ | r00f: you're a genius!!!!!!!! |
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| 08:37.52 | udp` | hi |
| 08:38.08 | udp` | At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/tempCallfile0.call |
| 08:38.14 | udp` | im getting this error |
| 08:38.20 | udp` | when im using |
| 08:38.33 | udp` | Application: Hangup() in my .call file |
| 08:38.37 | udp` | why is that? |
| 08:39.37 | Gugge | because at least one of app or extension must be specified, along with tech and dest. |
| 08:39.40 | Gugge | and you did not do that |
| 08:41.18 | Marquel | morning. when i send a fax call from a sip-client (ATA) the other side seems to misinterpret the call as data-call, while calling the same destination using a IAX client i can send my fax. have there been any changes or do i need to retest with other destinations? |
| 08:43.21 | udp` | Gugge: yeah i can also read that |
| 08:43.57 | udp` | Gugge: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out this guide says other thing |
| 08:44.22 | udp` | Gugge: maybe a reference to how do it correctly ? |
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| 08:49.39 | Gugge | udp`: without seeing the wrong file, its hard to help |
| 08:50.42 | udp` | Channel: SIP/Geils2/ |
| 08:50.44 | udp` | MaxRetries: 3 |
| 08:50.44 | udp` | RetryTime: 5 |
| 08:50.44 | udp` | WaitTime: 15 |
| 08:50.44 | udp` | Application: Hangup |
| 08:50.44 | udp` | data = ' ' |
| 08:51.09 | udp` | i deleted the number to call |
| 08:51.14 | Gugge | udp`: without seeing the wrong file, its hard to help |
| 08:51.18 | Gugge | and dont paste it here |
| 08:51.20 | Gugge | use a pastebin |
| 08:51.33 | udp` | well, this is the whole file |
| 08:55.22 | Gugge | så you want to call SIP/Geils2/, and when they pick up, you want to hangup? |
| 08:56.36 | Gugge | try the minimal call file example, start by changing the Channel to what you want, if that works, then try changing the app |
| 08:59.04 | Gugge | and remember to create the file somewhere else, and move it to outgoing |
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| 09:24.38 | udp` | Gugge: it is already working, the call is executed, but i dont know why i get warning message in the logs |
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| 10:40.03 | udp` | Unable to find a codec translation path from (g729) to (ulaw) |
| 10:40.08 | udp` | what is this error message? |
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| 10:45.31 | wdoekes | udp`: it probably means that someone is attempting to speak g729 and someone else speaks ulaw |
| 10:45.42 | wdoekes | asterisk would translate between the codecs |
| 10:45.49 | udp` | auhh ok thanx |
| 10:45.58 | wdoekes | but it doesn't have the g729 codec installed, so it can't |
| 10:46.05 | wdoekes | (you need a license) |
| 10:46.11 | udp` | yeah i just read that |
| 10:46.12 | udp` | :D |
| 10:46.43 | udp` | have you used .call files before ? |
| 10:47.58 | wdoekes | sure |
| 10:58.44 | udp` | i dont want to use extensions |
| 10:58.49 | udp` | in the .call files |
| 10:58.55 | udp` | so i have to use application |
| 10:59.19 | udp` | in the guides it says Hangup() |
| 10:59.23 | udp` | is an application |
| 10:59.40 | udp` | but when i use it says it is not exiest |
| 11:00.13 | udp` | wdoekes: i guess it is not exiest on my system |
| 11:00.26 | udp` | but why...? and how do i install it |
| 11:00.45 | wdoekes | udp`: back up a great deal |
| 11:00.50 | wdoekes | what do you want to achieve? |
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| 11:01.11 | udp` | im trying to do a stress test |
| 11:01.15 | udp` | making phone calls |
| 11:01.29 | udp` | and hangup if other side answer |
| 11:02.04 | wdoekes | ok. and if you type: asterisk -nrx 'core show application Hangup' |
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| 11:28.05 | meital_ | hi all. I finally managed to set up a server, and successfuly register 2 soft phones to it. But when I'm trying to call, I get "SIP/2.0 401 Unauthorized" error. Any suggestions? |
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| 11:33.27 | udp` | wdoekes: to the console ? |
| 11:35.16 | udp` | wdoekes: it gives me the specification of the function |
| 11:35.39 | wdoekes | 13:00 < udp`> wdoekes: i guess it is not exiest on my system |
| 11:35.46 | wdoekes | apparently it exists |
| 11:46.22 | udp` | wdoekes: so why do i get error message |
| 11:46.56 | wdoekes | I don't see any error message |
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| 11:53.23 | udp` | pbx.c: No such application 'Hangup()' |
| 11:53.37 | udp` | wdoekes: pbx.c: No such application 'Hangup()' |
| 11:54.12 | wdoekes | asterisk -nrx 'core show application Hangup()' |
| 11:55.47 | udp` | yes |
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| 11:57.03 | udp` | [Syntax] |
| 11:57.03 | udp` | Hangup([causecode]) |
| 12:03.24 | wdoekes | no |
| 12:03.31 | wdoekes | Hangup() != Hangup |
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| 12:43.31 | meital_ | someone? :( |
| 12:45.02 | [TK]D-Fender | What's the question? |
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| 12:52.16 | udp` | meital_: hi meital |
| 12:53.42 | meital_ | udp`: hi :) |
| 12:53.56 | meital_ | I finally managed to set up a server, and successfuly register 2 soft phones to it. But when I'm trying to call, I get "SIP/2.0 401 Unauthorized" error. Any suggestions? |
| 12:54.14 | meital_ | [TK]D-Fender: ^^ |
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| 12:55.00 | [TK]D-Fender | PASTEBIN the entire call for us with SIP DEBUG enabled |
| 12:55.01 | [TK]D-Fender | ~pb |
| 12:55.02 | 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. |
| 12:55.03 | [TK]D-Fender | ^^^ |
| 13:01.20 | meital_ | [TK]D-Fender: now I'm actually getting 404 not found -> http://pastebin.com/bVhUdpm1 |
| 13:03.19 | meital_ | [TK]D-Fender: the outout of tcpdump (where it says 301 Unauthorized) -> pastebin.com/mQiFkg7d |
| 13:06.11 | [TK]D-Fender | meital_: that is SIP OPTIONS. That is not a call |
| 13:06.27 | meital_ | [TK]D-Fender: what do you mean? |
| 13:06.38 | [TK]D-Fender | what part of "that is not a call" is unclear? |
| 13:07.34 | [TK]D-Fender | the 404 is not a call |
| 13:07.42 | meital_ | [TK]D-Fender: lol...it happens when I initiat a call from android application,: |
| 13:07.44 | meital_ | call = manager.makeAudioCall(me.getUriString(), sipAddress, listener, 30); |
| 13:07.49 | meital_ | so it is a call... |
| 13:08.06 | [TK]D-Fender | [09:01]meital_[TK]D-Fender: now I'm actually getting 404 not found -> http://pastebin.com/bVhUdpm1 <- this is not a call |
| 13:08.16 | [TK]D-Fender | Show us Asterisk SIP debug for a call attempt |
| 13:09.42 | [TK]D-Fender | http://pastebin.com/mQiFkg7d <- this also has no calls in it. It has failed registration attempts. |
| 13:09.50 | [TK]D-Fender | <PROTECTED> |
| 13:09.53 | [TK]D-Fender | with* |
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| 13:13.20 | meital_ | [TK]D-Fender: no i'm trying from a soft phone, and I see "Call from '2345' (192.168.10.84:52420) to extension '1234' rejected because extension not found in context 'demo'." |
| 13:13.54 | meital_ | it seems like a configuration issue in sip.conf right? |
| 13:13.59 | [TK]D-Fender | that error is not in the 2 pastebins your provided |
| 13:14.04 | [TK]D-Fender | no, that is a DIALPLAN error |
| 13:14.16 | [TK]D-Fender | EXTENSIONS.CONF |
| 13:14.28 | [TK]D-Fender | You have not configured your dialplan to process what was dialed. |
| 13:15.14 | meital_ | [TK]D-Fender: http://pastebin.com/urcEAxSf |
| 13:15.41 | meital_ | http://pastebin.com/mtGW5yZL |
| 13:16.35 | [TK]D-Fender | That error means exactly what it says. |
| 13:16.43 | [TK]D-Fender | "Call from '2345' (192.168.10.84:52420) to extension '1234' rejected because extension not found in context 'demo'." |
| 13:16.53 | [TK]D-Fender | No match for '1234' |
| 13:17.31 | meital_ | but there is a match (look at http://pastebin.com/urcEAxSf) |
| 13:17.38 | [TK]D-Fender | There isn't. |
| 13:17.47 | [TK]D-Fender | not found in context 'demo' <--------------------------- |
| 13:18.32 | meital_ | so I really don't know how to configure it...shouldn't the context be defined only on sip.conf? |
| 13:18.52 | [TK]D-Fender | meital_: your peer's point to a context in extensions.conf |
| 13:19.01 | [TK]D-Fender | You pointed them to one that does not exist at all |
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| 13:20.20 | [TK]D-Fender | http://pastebin.com/mtGW5yZL <- 9, 17, 26. You told these devices to send their calls into [demo] in extensions.conf. It does not exist |
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| 13:27.07 | meital_ | [TK]D-Fender: ok thanks, it's working now from the soft phone. I'll try to figure out why it's still not working from the android app |
| 13:31.33 | [TK]D-Fender | When you've got something, show us. |
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| 13:34.33 | *** join/#asterisk fiderosado (~fidel@200.55.157.220) |
| 13:34.58 | fiderosado | :-Dhello |
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| 13:36.29 | fiderosado | Hi I need someone who can help me to configure asterisk on my pc |
| 13:37.04 | meital_ | [TK]D-Fender: I restarted the application on both android devices, no I get 603 declined instead. Which conf file do I need to take a look at? |
| 13:37.48 | meital_ | *now |
| 13:38.19 | fiderosado | someone who speaks Spanish around here |
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| 13:38.29 | anonymouz666 | que pasa? |
| 13:38.36 | [TK]D-Fender | meital_: Don't ask "what confi". 603 could be a response to all sorts of things. Show us the call. |
| 13:39.35 | fiderosado | anonymouz666 sabes cnfigurar asterizk |
| 13:40.18 | anonymouz666 | fiderosado: estoy aprendiendo |
| 13:40.27 | fiderosado | ya tu sabes jeje |
| 13:40.32 | fiderosado | y que has logrado hacer |
| 13:40.51 | fiderosado | porke yo empece anoche ccon esa app |
| 13:40.58 | fiderosado | y nada se |
| 13:41.29 | fiderosado | solo hice unos cambis y una pruebas y creo que me esta funcinando |
| 13:41.50 | fiderosado | per necesito saber cmo funcionaria crrectamente |
| 13:42.06 | meital_ | [TK]D-Fender: how can I identify that there is a call in the log that I'm sending you, so I won't send something wrong again? |
| 13:42.25 | [TK]D-Fender | meital_: Do you see the 603 in there? Do you see an INVITE from your device? |
| 13:42.32 | anonymouz666 | fiderosado: lees el libro? |
| 13:42.52 | fiderosado | si me lei el readme que trae |
| 13:43.08 | fiderosado | pero necesito un buen tutorial |
| 13:43.17 | anonymouz666 | the book! |
| 13:43.36 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-vhpgydvexexfnifc) |
| 13:43.50 | fiderosado | no se cual seria ese libro entonces |
| 13:44.14 | anonymouz666 | ~book |
| 13:44.15 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:45.20 | meital_ | [TK]D-Fender: yes, I can see 603 here -> http://pastebin.com/z1eU9Cg4 |
| 13:45.42 | dw5304 | wo i asked this one before but ill ask again t1 card should i buy it with or with out a echo cancel module |
| 13:46.16 | WIMPy | dw5304: Some days ago? |
| 13:46.20 | dw5304 | yea |
| 13:46.21 | [TK]D-Fender | meital_: "I see no verbose and that doesn't include the start of the call |
| 13:46.36 | dw5304 | trying to get a feal from multiple people |
| 13:46.37 | dw5304 | <PROTECTED> |
| 13:46.45 | WIMPy | dw5304: I haven't changed my mind since then :-) |
| 13:46.53 | [TK]D-Fender | dw5304: With |
| 13:47.25 | dw5304 | see this is my issue :) its allways a 50/50 |
| 13:48.07 | fiderosado | no tngo acceso al libro anonymouz666 |
| 13:49.16 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xyjxwhpvlzwviwif) |
| 13:49.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:50.16 | meital_ | [TK]D-Fender: I used "sip set debug on", should I use something else? How can I tell where is the beggining of the call? I really appriciate your help! I started using asterisk only yesterday... |
| 13:50.30 | [TK]D-Fender | meital_: You don't have the very start of the call in there. |
| 13:51.10 | [TK]D-Fender | meital_: :"core set verbose 10" <- I saw no dialplan execution in there either. Ensure your verbose is high enough. Go place another call |
| 13:51.16 | apeer | How to enable DTLS-SRTP support in Asterisk 11 for a WebRTC call to Chrome 35 |
| 13:52.19 | anonymouz666 | apeer: Chrome changes does not mean that Asterisk will blind follow |
| 13:52.32 | anonymouz666 | I have this issue with Chrome 35 |
| 13:52.43 | apeer | chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 53226 UDP/TLS/RTP/SAVPF |
| 13:52.52 | anonymouz666 | yes, exactly this one |
| 13:53.18 | apeer | I'm struggling for 2 days now. |
| 13:53.42 | anonymouz666 | apeer: let's say you make this work with Chrome 35 somehow, now imagine the 36 changed the handshake again. |
| 13:53.56 | anonymouz666 | how can you trust on this? |
| 13:54.00 | fiderosado | <PROTECTED> |
| 13:54.22 | anonymouz666 | that's why I stopped with webrtc for while |
| 13:54.31 | apeer | what is the use of WebRTC then. |
| 13:55.00 | file | WebRTC isn't finished. |
| 13:55.05 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:55.05 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:55.21 | anonymouz666 | file is finishing it :P |
| 13:55.43 | file | I did update support in Asterisk 11 branch as of yesterday to fix it up |
| 13:55.52 | file | but do I think that'll be the only changes required in the end? no |
| 13:57.35 | file | that being said - using the Asterisk 11 branch you can establish calls but it requires a DTLS capable OpenSSL and you also have to generate certificates to use for the DTLS negotiation |
| 14:01.25 | meital_ | [TK]D-Fender: http://pastebin.com/G4jsrgQt |
| 14:05.18 | apeer | file: can you do DTLS with a self assigned certificate (just as it is documented for tls? |
| 14:05.26 | file | sure |
| 14:05.27 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ypfkfqmcqvpsdstz) |
| 14:05.27 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:05.39 | file | just don't enable certificate verification and it'll work fine |
| 14:06.15 | [TK]D-Fender | [09:51][TK]D-Fendermeital_: :"core set verbose 10" <- I saw no dialplan execution in there either. Ensure your verbose is high enough. Go place another call |
| 14:06.29 | [TK]D-Fender | again you have no verbose enabeld |
| 14:07.10 | file | Er Disclaimer: That doesn't come with a guarantee. |
| 14:08.25 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:08.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.33 | fiderosado | hola alguien con experiencia que me ayude a configurar asterisk / hello someone with experience to help me configure asterisk |
| 14:10.13 | anonymouz666 | fiderosado: you need to make specific questions |
| 14:10.26 | anonymouz666 | generic questions gives to you generic answers |
| 14:10.30 | fiderosado | echarl only need to run under a domain |
| 14:10.48 | fiderosado | only need to run under a domain |
| 14:11.32 | fiderosado | I think it's in sip.conf but do not know much |
| 14:14.24 | meital_ | [TK]D-Fender: http://pastebin.com/VWpcUdDB |
| 14:15.31 | meital_ | fiderosado: I suggest that you google "asterisk basic config" |
| 14:15.48 | meital_ | there are a few files to edit |
| 14:16.06 | [TK]D-Fender | meital_: 648 <- your ran out of dialplan to execute and you never answered the call and didn't have any other status to pass on. |
| 14:16.08 | fiderosado | ;Use "sip show domains" to list local domains |
| 14:16.08 | fiderosado | domain = sip1.domain.net,domain.net,192.168.0.2 |
| 14:16.48 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-edvwyihqywveaout) |
| 14:17.40 | fiderosado | ok |
| 14:17.44 | fiderosado | i see |
| 14:21.56 | *** join/#asterisk michael_work (~michael@bzq-82-168-31-134.red.bezeqint.net) |
| 14:21.56 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 14:22.21 | michael_work | hey |
| 14:23.52 | michael_work | When i trigger Originate from AMI i as well send Data. At some point i added to much data and i'm getting "the line is too long message" |
| 14:24.14 | michael_work | currently i send data as one line |
| 14:24.42 | *** join/#asterisk Yon (~Yon@c80-217-134-166.bredband.comhem.se) |
| 14:24.43 | michael_work | is there way to break it in pieces or i'll need to use AGI :( |
| 14:24.43 | [TK]D-Fender | Which is apparently too long |
| 14:25.03 | [TK]D-Fender | You should probably show us what you're doing... |
| 14:26.10 | *** join/#asterisk salz212 (~chatzilla@li345-39.members.linode.com) |
| 14:27.13 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:27.14 | michael_work | my problem that the Variable i'm sending is one long line |
| 14:28.24 | [TK]D-Fender | There is a limit on dialplan processing of a var that big as well, but you might get away with sending it as 2 pieces and combining them in your dialplan |
| 14:29.04 | michael_work | the line contains several variables with delimeter |
| 14:30.00 | [TK]D-Fender | Why are you not jsut setting each individiually? |
| 14:30.31 | michael_work | [TK]D-Fender, meaning? |
| 14:30.50 | [TK]D-Fender | 1 line each |
| 14:31.03 | michael_work | manager show command Originate |
| 14:31.15 | [TK]D-Fender | I'm familiar the with the command. |
| 14:31.18 | [TK]D-Fender | Show use your packet |
| 14:31.30 | michael_work | <PROTECTED> |
| 14:31.32 | michael_work | good |
| 14:31.34 | michael_work | thanks |
| 14:31.37 | michael_work | stupid me |
| 14:31.46 | [TK]D-Fender | multiple Variable: headers are allowed <- |
| 14:31.48 | [TK]D-Fender | yup :p |
| 14:31.49 | michael_work | i saw only the name and did not see that multiple headers allowed |
| 14:32.03 | michael_work | so i though maybe there is way to send one but with multiple lines |
| 14:32.12 | michael_work | i'll fix our lib working with Asterisk |
| 14:32.38 | michael_work | as i do not want to keep them in memory and use AGI to get them to channel :P |
| 14:33.10 | [TK]D-Fender | multiple lines..... |
| 14:33.23 | michael_work | [TK]D-Fender, thanks, though i'm embarassed now that i missed it :( |
| 14:33.43 | [TK]D-Fender | michael_work: No biggie... |
| 14:33.45 | michael_work | multiple headers != multiple lines :) |
| 14:38.36 | apeer | @file used these settings:udpbindaddr=0.0.0.0:5060 |
| 14:38.36 | apeer | realm=213.148.146.118 |
| 14:38.36 | apeer | transport=udp,ws,wss |
| 14:38.36 | apeer | dtlsenable=yes |
| 14:38.36 | apeer | dtlsverify=no |
| 14:38.36 | apeer | dtlscertfile=/etc/asterisk/keys/asterisk.pem |
| 14:38.37 | apeer | dtlscafile=/etc/asterisk/keys/ca.crt |
| 14:38.37 | apeer | dtlscipher=ALL |
| 14:38.38 | apeer | dtlssetup=actpass |
| 14:39.26 | apeer | with a svn checkout branches 11 still no luck |
| 14:39.36 | michael_work | apeer, what are you trying? |
| 14:39.48 | apeer | webrtc |
| 14:39.52 | michael_work | dtls? |
| 14:39.56 | apeer | yes |
| 14:40.18 | michael_work | i tried with 11.5 i think |
| 14:40.25 | michael_work | and it worked |
| 14:40.35 | michael_work | not dtls though |
| 14:40.58 | [TK]D-Fender | apeer: PASTEBIN. Do not flood in here. |
| 14:41.00 | apeer | chrome 35 seems the problem and it was supposed to be fixed yesterday |
| 14:41.09 | file | I can't help at all without console output, information about browser, what javascript library you're using, etc |
| 14:41.32 | michael_work | chrome has problem with one way audio at certain computers :( at least at my latests checks but it worked |
| 14:42.07 | michael_work | are there logs of chan btw? |
| 14:42.20 | michael_work | i would like to read, as i'm gonna poke webrtc soon again |
| 14:43.51 | file | I just called from a Chrome derivative into an Asterisk 12 using chan_pjsip |
| 14:46.09 | *** join/#asterisk u0m3 (~u0m3@92.80.103.49) |
| 14:46.35 | meital_ | [TK]D-Fender: It's working!!! thank you so much!!! (start to the android solved it) |
| 14:47.29 | [TK]D-Fender | meital_: Not sure what your Android issue was there.. but that last call was just "ended with not status to pass on and no answer" |
| 14:51.07 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-yxspxxhqumzjgblu) |
| 14:58.14 | apeer | webrtc I'm using sipml5.org site to test here are is the log http://pastebin.com/pajbKW4g |
| 14:58.54 | file | put force_avp=yes in your SIP entry |
| 14:59.51 | file | (and remember to reload it) |
| 15:01.45 | file | and put the DTLS config in the SIP entry, I can't remember if it works in general |
| 15:01.49 | file | (I don't think so) |
| 15:04.26 | *** join/#asterisk r00f (~r00f@94.204.12.229) |
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| 15:04.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:05.07 | apeer | this is the sip entry now http://pastebin.com/6rsZhht2 |
| 15:07.57 | apeer | @file: thank you, no audio yet but Executing [1234@from-internal:1] Playback("SIP/6001-00000001", "demo-congrats") in new stack |
| 15:12.07 | *** join/#asterisk wonderworld (~ww@p5084BF0F.dip0.t-ipconnect.de) |
| 15:27.27 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 16:11.58 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 17:19.32 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 17:21.57 | dan_j | Hi, some sip peers keep receiving calls from random callers. These calls arent originating from that asterisk server, but are (i assume) direct ip calls. |
| 17:22.09 | dan_j | This has been happening with both Polycom and Yealink phones. |
| 17:22.18 | dan_j | Does anyone have experience at blocking these calls? |
| 17:22.44 | dan_j | So far, i've been changing the external port number from 5060 to something else, but for one client, that doesnt appear to have helped. |
| 17:22.58 | WIMPy | Why are they reachable at all? |
| 17:23.20 | dan_j | Do you know the name of a setting which i should be looking for? |
| 17:24.08 | dan_j | WIMPy: the sip peers are behind a nat device. I assume that the caller is sending sip invites to the external nat port and the rubbish router is allowing the call through, even if its not coming from the correct source ip. |
| 17:24.32 | WIMPy | Accept calls from registrar only? |
| 17:24.48 | dan_j | You would expect it to be called that :-S |
| 17:25.01 | WIMPy | That sound like some rather strange behaviour for NAT. |
| 17:25.14 | *** join/#asterisk fiderosado (~fidel@200.55.157.235) |
| 17:25.24 | dan_j | I've had these kind of issues quite often. Usually when the external nat port happens to be 5060. |
| 17:26.15 | WIMPy | I haven't seen such a thing happening. |
| 17:26.37 | fiderosado | hello |
| 17:26.41 | dan_j | It seems to happen to around 2 in 5 of my clients. |
| 17:26.49 | WIMPy | Could that be some rather unwanted UPNP magic? |
| 17:27.11 | dan_j | Hmm. maybe. |
| 17:33.47 | dan_j | It seems that the setting on Yealinks is hidden in the provisioning files. |
| 17:33.48 | dan_j | http://forum.yealink.com/forum/showthread.php?tid=1418 |
| 17:39.04 | fiderosado | me pueden decir si es posible configurar asterisk para que marque el puerto del modem que tengo en mi pc para acceder a la linea de telefnia basica |
| 17:42.04 | fiderosado | I can tell if it is possible to configure asterisk to dial the modem port I have on my pc to access to basic telephony |
| 17:43.40 | [TK]D-Fender | fiderosado: No. Generic modems are not supported by Asterisk |
| 17:50.32 | fiderosado | and what type of modem that I used to |
| 17:53.15 | [TK]D-Fender | fiderosado: FORGET modems |
| 17:53.35 | [TK]D-Fender | those are not happening. You need a supported FXO interface like Digium & Sangoma's cards |
| 17:53.40 | fiderosado | <PROTECTED> |
| 17:53.55 | [TK]D-Fender | modem = useless |
| 17:54.20 | WIMPy | ... with Asterisk >1.0 |
| 17:54.39 | fiderosado | I do not understand what you wear, I apologize but I'm new at this |
| 17:54.48 | *** join/#asterisk MrJoshGeddes (storm3y@unaffiliated/storm3y) |
| 17:55.10 | WIMPy | Modem support was removed from Asterisk with version 1.2. |
| 17:55.30 | fiderosado | if I get a card or interface can serve me digium |
| 17:57.30 | fiderosado | I use the PBX application manager free edition version 2.0 build 2007.01.30 |
| 17:57.34 | Katty | he wears his sunglasses at night. |
| 17:57.53 | WIMPy | fiderosado: What is that? |
| 17:58.14 | fiderosado | i dont now jejee |
| 17:58.38 | fiderosado | is asterisk for windows |
| 17:59.09 | Katty | it's for cats. |
| 17:59.13 | WIMPy | Uh. That has beed dead for ages as far as I know. |
| 17:59.32 | WIMPy | For cats or for Kattys? |
| 17:59.38 | fiderosado | jeje |
| 17:59.54 | Katty | why not both?! |
| 18:01.23 | fiderosado | I know it's alg n to what you are used to working, because I have a machine with linux repository and no hand |
| 18:01.58 | fiderosado | and my internet access is limited in speed |
| 18:02.12 | Katty | hands are a key feature of asterisk. |
| 18:02.30 | *** join/#asterisk MarcoZink (~marcozink@201.149.45.172) |
| 18:03.02 | fiderosado | good to see those words, thank katty |
| 18:04.25 | fiderosado | Recommend me sque operating system to mount server with asterisk knowing the characteristics of my network |
| 18:04.46 | fiderosado | repeat - Recommend me which OS to mount server with asterisk knowing the characteristics of my network |
| 18:05.23 | [TK]D-Fender | fiderosado: Any standard Linux should work fine. |
| 18:05.44 | Katty | i like the Hands OS, personally. |
| 18:06.19 | [TK]D-Fender | 's PBX touched him a "bad" place. |
| 18:07.06 | fiderosado | katty and you have something mounted asterisk |
| 18:07.37 | Katty | no. i keep all the boys away from my asterisk servers. |
| 18:07.40 | fiderosado | because I need some experience to guide me. lol I'm new to this |
| 18:08.51 | fiderosado | my goal is to learn not to hurt anyone |
| 18:08.58 | mjordan | Have you read Asterisk: The Definitive Guide? |
| 18:09.01 | mjordan | ~book |
| 18:09.01 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:09.09 | mjordan | (you probably should) |
| 18:09.51 | fiderosado | <PROTECTED> |
| 18:11.36 | [TK]D-Fender | ...because? |
| 18:11.44 | fiderosado | I was thinking that if linking a database mysql with asterisk server I can add users more quickly, it would be possible to set |
| 18:12.08 | [TK]D-Fender | fiderosado: read the book. |
| 18:12.13 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 18:12.15 | fiderosado | <PROTECTED> |
| 18:12.30 | [TK]D-Fender | use a proxy site then |
| 18:13.00 | file | oh |
| 18:13.04 | file | grins |
| 18:13.12 | fiderosado | if I use a webproxy the internet connection gets me too slow, but will try to download |
| 18:13.21 | file | try http://rho.jcn-network.com/asteriskdocs.org/ |
| 18:13.38 | Katty | look at file. |
| 18:13.42 | Katty | he's practically the ball. |
| 18:14.04 | file | bounces |
| 18:14.06 | fiderosado | yes http://rho.jcn-network.com/asteriskdocs.org/ is very fast for me |
| 18:14.20 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 18:14.36 | *** join/#asterisk hecatae (~philip@host-92-28-13-204.as13285.net) |
| 18:15.04 | file | there you go, the book |
| 18:15.11 | fiderosado | as I can download that book. |
| 18:19.41 | fiderosado | lol copy and send all pages of references |
| 18:20.47 | *** part/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 18:20.50 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 18:20.51 | *** mode/#asterisk [+o file] by ChanServ |
| 18:21.00 | file | Katty, hi |
| 18:21.18 | *** join/#asterisk MrJoshGeddes (storm3y@unaffiliated/storm3y) |
| 18:22.42 | Katty | file: herro. |
| 18:22.43 | fiderosado | any of ustes can help me configure asterisk least explain how I have to configure users and that of outgoing extensions, you do not understand anything lol |
| 18:23.01 | file | Katty, happy Canada day! |
| 18:23.16 | fiderosado | repeat - any of you can help me to configure asterisk least explain how I have to configure users and that of outgoing extensions, you do not understand anything lol |
| 18:23.25 | Katty | file: thank you, my fine canadian friend! |
| 18:23.33 | Katty | file: i hope you're celebrating with poutine |
| 18:23.45 | file | Katty, I am not - I am celebrating with air conditioning |
| 18:24.29 | [TK]D-Fender | fiderosado: Noone is going to try to explain all about asterisk to you here over IRC across a language barrier. |
| 18:26.53 | fiderosado | [TK]D-Fender , I just need to know some basic things, you know that the extensions as creating accounts just that, and that aid growth is the main motivation that a developer should have |
| 18:28.45 | [TK]D-Fender | fiderosado: Your wording is almost incomprehensible |
| 18:30.37 | newtonr | fiderosado, I can't understand you either. You might try looking at https://wiki.asterisk.org/wiki/display/AST/Hello+World |
| 18:30.40 | fiderosado | is to use google translate translate my words that may be a little out of context, but my English is not very good understand them perfectly |
| 18:32.11 | Katty | file: why not both? |
| 18:32.32 | file | Katty, perhaps in the future... perhaps |
| 18:36.17 | mjordan | fiderosado: I would try working through Asterisk: The Definitive Guide and ask specific questions when you hit something you don't understand. Your questions are too generic for anyone to accurately answer here. |
| 18:38.06 | mic_ | anyone using DINSTAR gsm gateways? |
| 18:39.01 | Katty | i would like to know WHY the SHA-256 branch was fixed so quickly. |
| 18:39.13 | Katty | RE: DTLS. |
| 18:39.17 | fiderosado | ok mjordan're right, just download the book just reading it I'll see if I get many doubts |
| 18:39.56 | Katty | actually it wasn't that quick. |
| 18:40.23 | file | Katty, what? |
| 18:40.39 | Katty | exactly. |
| 18:41.10 | mjordan | two weeks of file's time is a looooong time. |
| 18:41.30 | fiderosado | byy |
| 18:41.37 | fiderosado | logout |
| 18:42.15 | Katty | i would like to know what ticketing system you guys use. |
| 18:42.34 | fiderosado | quit |
| 18:42.44 | file | the Asterisk project uses JIRA |
| 18:43.16 | Katty | the godzilla ticketing system. |
| 18:43.31 | Katty | is it hosted for you, or in house? |
| 18:43.34 | file | mjordan would call it other things |
| 18:43.42 | file | in house |
| 18:43.45 | Katty | and i'm sure they'd all be colorful. |
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| 19:16.45 | ChannelZ | OT I know but anyone overly familiar with PCIe power connector pinouts? |
| 19:17.03 | ChannelZ | Or really, the connectors |
| 19:17.30 | WIMPy | wikipedia |
| 19:18.51 | ChannelZ | yes but so far I haven't been able to find any pictures of the power connectors. |
| 19:19.39 | ChannelZ | Basically I got a 2x4pin molex to 6-pin PCIe adapter (or what should be) which, on the 6-pin side, are normally one row that is TTS and the next STT where T is tapered and S is square. |
| 19:20.26 | ChannelZ | But this one seems to be TSS and STT. I've not encountered this before.. I don't know if it's a manufacturing mistake, or what that particular connector would be used for. |
| 19:21.12 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
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| 19:35.03 | hecatae | ChannelZ: I fell like I'm reading a very foreign language |
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| 19:39.49 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:47.54 | apeer | :@file I have a webrtc call but no audio, streams are running please see log http://pastebin.com/2HG34BUM |
| 19:48.20 | file | that does not help - I need sip debug and rtp debug |
| 19:50.20 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 19:55.06 | apeer | :@file rtp debug : http://pastebin.com/1bvD4qp1 |
| 19:55.43 | file | looks like ICE negotiated fine, you'd have to look at the Chrome debug log to see what it is saying |
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| 19:58.58 | apeer | :@file sip debug http://pastebin.com/wRgUBdMF |
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| 19:59.35 | file | yeah signaling is fine |
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| 20:08.53 | apeer | :@file the chrome debug console http://pastebin.com/htkP0xbG |
| 20:09.21 | file | that's not the chrome debug log |
| 20:10.21 | file | and that's the extent of WebRTC for me today |
| 20:14.34 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 20:15.01 | Kobaz | i'm a little sip trunk, short and stout |
| 20:15.11 | Kobaz | here's my address... right in the cloud |
| 20:16.47 | mic_ | teh cloud. your highway to distributed failure. |
| 20:16.53 | mic_ | <PROTECTED> |
| 20:16.54 | mic_ | ;) |
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| 20:32.26 | Kobaz | private-cloud is good |
| 20:32.26 | Kobaz | public cloud places too much faith in lots of third parties |
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| 20:33.39 | ipengineer | If I want to do a h.264 video call between two endpoints what kind of load will this put on Asterisk say compared to a G711 voice call since they are both the same codec? |
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| 20:37.55 | [TK]D-Fender | cirtually none as * only repacks the media |
| 20:38.07 | [TK]D-Fender | since * can't transcode video. |
| 20:38.15 | [TK]D-Fender | BBIAB |
| 20:47.46 | apeer | :@file debug log chrome http://pastebin.com/h1cJv6aq |
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| 21:31.18 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 22:10.38 | *** join/#asterisk portalBlock (~portalBlo@2607:5300:60:2d42::1) |
| 22:12.32 | portalBlock | Hello, I have a very small budget and need to setup a PBX, I have 2 modems here that are larger then normal and they both say telco(port 1) and phone(port 2). What are the chances they will work? (The reason I am asking is because my friend had some success with a regualr modem a few years ago) |
| 22:12.52 | WIMPy | 0 |
| 22:13.01 | portalBlock | I am aware that you need fxs/fxo |
| 22:13.04 | portalBlock | Oh.. |
| 22:13.12 | Qwell | WIMPy: possibly less |
| 22:13.53 | dlynes_laptop | portalBlock: Unless you have analog/pri passthrough, what Qwell said |
| 22:14.01 | [TK]D-Fender | Qwell: It would be less but they made it an unsigned int... |
| 22:14.09 | Qwell | [TK]D-Fender: still less |
| 22:14.13 | TazzNZ | so...has any of the long time asterisk dev's here ever *seen* Asterisk work with a Voice Modem from the good old days ? |
| 22:14.16 | WIMPy | dlynes_laptop: What? |
| 22:14.22 | Qwell | TazzNZ: sure, I own several |
| 22:14.42 | portalBlock | What version works with voice modems? |
| 22:14.47 | Qwell | none |
| 22:14.49 | WIMPy | TazzNZ: Asterisk 1.0 supprted them. |
| 22:14.53 | portalBlock | Oh. |
| 22:15.03 | TazzNZ | wow....I didn't know that :) |
| 22:15.05 | Qwell | WIMPy: it supported one, and only one. |
| 22:15.10 | TazzNZ | lol |
| 22:15.13 | TazzNZ | *one* |
| 22:15.15 | TazzNZ | :D |
| 22:15.17 | [TK]D-Fender | well... 2 actually |
| 22:15.23 | Qwell | [TK]D-Fender: quiet, you |
| 22:15.26 | WIMPy | Only one? |
| 22:15.33 | WIMPy | Were they special? |
| 22:15.38 | Qwell | WIMPy: nope |
| 22:15.49 | [TK]D-Fender | WIMPy: in an olympic kind of way, yes |
| 22:15.55 | Qwell | [TK]D-Fender: ...special |
| 22:16.01 | portalBlock | So what can I do? When I say small budget I mean we have the computer but need a way to get PSTN to it and I only have about $25-$50 to work with. |
| 22:16.01 | WIMPy | Then why wouldn;t any modem have worked? |
| 22:16.14 | Qwell | WIMPy: because there were only drivers written for that one. |
| 22:16.33 | WIMPy | portalBlock: Make a human phone to IRC gateway. |
| 22:16.39 | [TK]D-Fender | portalBlock: portalBlock How many lines do you need to take in? |
| 22:16.45 | portalBlock | [TK]D-Fender, 1 |
| 22:16.52 | TazzNZ | Qwell: how did you get the audio into asterisk ? |
| 22:16.52 | WIMPy | But teh modems were pretty mch standard. |
| 22:16.55 | [TK]D-Fender | portalBlock: Linksys SPA-3102 <--- |
| 22:17.07 | Qwell | WIMPy: you'd think so |
| 22:17.20 | portalBlock | Where can I get one that I dont have to worry about it being locked? |
| 22:17.26 | portalBlock | (I hear that somewhere) |
| 22:17.31 | [TK]D-Fender | portalBlock: I've never seen these locked ever |
| 22:17.32 | portalBlock | s/hear/heard |
| 22:17.37 | dlynes_laptop | portalBlock: 3102's have never been blocked |
| 22:17.38 | portalBlock | Oh. Good :D |
| 22:17.45 | WIMPy | That was the nice thing when I changed from POTS to ISDN. My voicemail still worked with the new hardware. |
| 22:17.45 | dlynes_laptop | portalBlock: you're confusing them with 2102's |
| 22:17.53 | portalBlock | Ohh |
| 22:18.06 | WIMPy | That was pre Asterisk off course. |
| 22:18.31 | portalBlock | Also, to forward and incoming call to an external phone number do I need more then 1 PSTN line? |
| 22:18.42 | dlynes_laptop | portalBlock: yes |
| 22:18.47 | portalBlock | Aww |
| 22:19.01 | WIMPy | Fortward as in forward or forward as in transfer? |
| 22:19.04 | dlynes_laptop | portalBlock: unless your analog service provider provides a call forwarding service |
| 22:19.06 | portalBlock | transfer |
| 22:19.29 | portalBlock | dlynes_laptop, they do, but for a fee as per normal (remeber I said budget) |
| 22:19.30 | WIMPy | Then you most probably need two lines per call. |
| 22:19.34 | portalBlock | Ok |
| 22:19.57 | portalBlock | WIMPy, whats the difference between forward/transfer? |
| 22:20.18 | WIMPy | forwarding happens in the switch. |
| 22:20.45 | WIMPy | So can transfers. But I don't know any telco that enables that feature. |
| 22:21.03 | Qwell | Is there any reason you need to stick with analog? It sounds like you're spending more per month on that line, that your entire PBX budget. |
| 22:21.04 | dlynes_laptop | WIMPy: they typically only enable that feature for centrex lines |
| 22:21.09 | WIMPy | They usually can't find out how to bill it. |
| 22:21.10 | [TK]D-Fender | WIMPy: I believe I've seen one. Ever. |
| 22:23.06 | portalBlock | Qwell, thats all thats offered around here. |
| 22:23.19 | Qwell | You don't have the Internet where you are? |
| 22:23.23 | WIMPy | emigrate |
| 22:23.33 | TazzNZ | lol WIMPy |
| 22:25.27 | portalBlock | Can the spa-3102 act as FXO *and* FXS? |
| 22:25.58 | portalBlock | (Have the line come in and a phone hooked up as an extension) |
| 22:26.26 | portalBlock | Qwell, we do but we like to support local businesses and that is just not an option right now. |
| 22:26.33 | [TK]D-Fender | oiryes |
| 22:26.39 | [TK]D-Fender | portalBlock: yes |
| 22:26.47 | portalBlock | :D Where can I get one? |
| 22:26.57 | [TK]D-Fender | portalBlock: Places that sell them |
| 22:27.19 | portalBlock | Well yeah... but what stores would sell them? |
| 22:27.30 | portalBlock | Would radio shack? |
| 22:27.34 | [TK]D-Fender | I have a local computer store taht does... |
| 22:27.41 | [TK]D-Fender | no, radio shack would not.... |
| 22:27.42 | Qwell | SPA-3012 does not contain batteries, so no. |
| 22:27.49 | [TK]D-Fender | that is not a "consumer device" by most accounts |
| 22:28.11 | [TK]D-Fender | Welcome to Radio Shack, you have questions ... we have batteris! |
| 22:28.32 | Qwell | [TK]D-Fender: "You've got questions, we've got dumbfounded expressions." |
| 22:28.34 | portalBlock | Lol, not a fan of radio shack here :P |
| 22:28.43 | [TK]D-Fender | portalBlock: they changed.... |
| 22:28.52 | portalBlock | Humm? |
| 22:28.56 | [TK]D-Fender | portalBlock: Somewhere around the time of the introduction of the DVD player |
| 22:29.07 | portalBlock | Uhh, ok? |
| 22:30.44 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 22:32.22 | *** join/#asterisk Draecos (~Draecos@101.112.4.188) |
| 22:33.50 | [TK]D-Fender | http://www.newegg.com/Product/Product.aspx?Item=N82E16833150194&cm_re=SIP_FXO-_-33-150-194-_-Product |
| 22:34.05 | [TK]D-Fender | http://www.newegg.com/Product/Product.aspx?Item=N82E16833617002&cm_re=SIP_FXO-_-33-617-002-_-Product |
| 22:36.34 | portalBlock | [TK]D-Fender, first one has way too many features :P but thanks |
| 22:41.47 | portalBlock | I take it this is it? http://www.amazon.com/Cisco-SPA3102-Voice-Gateway-Router/dp/B000FKP55U |
| 22:43.03 | [TK]D-Fender | yes |
| 22:43.13 | portalBlock | Cool. |
| 22:46.04 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 22:46.31 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 22:46.50 | portalBlock | Forgot to ask, what are the system requirments for asterisk? |
| 22:46.59 | ageis | I can't figure out how to register this Polycom... |
| 22:47.01 | ageis | [Jul 1 22:45:17] WARNING[7437]: chan_sip.c:16405 check_auth: username mismatch, have <conf>, digest has <> |
| 22:47.07 | ageis | [Jul 1 22:45:17] NOTICE[7437]: chan_sip.c:28003 handle_request_register: Registration from '<sip:conf@voip.example.com>’ failed for '75.144.*.*:1024' - Username/auth name mismatch |
| 22:48.43 | portalBlock | I have a intel core2duo with 6GB (going to be 8GB soon) ram. it is going to be running asterisk(freepbx distro) virtualized and a DNS server, is that good? |
| 22:49.31 | TazzNZ | I would advise against running Asterisk in VM's |
| 22:50.21 | portalBlock | Uhh, thats not great for our setup. May I ask why? |
| 22:50.35 | WIMPy | But if your budget is an issue, you could sell the RAM. |
| 22:50.43 | TazzNZ | lol |
| 22:51.21 | portalBlock | "<portalBlock> Forgot to ask, what are the system requirments for asterisk?" Thats why I am asking... |
| 22:51.24 | TazzNZ | portalBlock: if you need Asterisk (or any realtime application) to do any realtime work, a VM will not work |
| 22:51.28 | WIMPy | The last Asterisk bo I mad ran on a Celeron with 128MB. |
| 22:51.39 | TazzNZ | since the CPU time is assigned |
| 22:51.48 | TazzNZ | not fixed, like a physical machine |
| 22:51.59 | WIMPy | But as long as you only do VOIP, that's not realtime. |
| 22:52.10 | portalBlock | Mmk, good to know. Guess I will pull out this old Dell vostro 1500 and use it. |
| 22:52.23 | TazzNZ | you could also run DNS and Asterisk on the same box |
| 22:52.52 | portalBlock | TazzNZ, I could... but I am not sure how/if the FreePBX distro will like that. |
| 22:53.04 | TazzNZ | it shouldn't care |
| 22:53.09 | TazzNZ | as it doesn't manage DNS |
| 22:53.35 | portalBlock | Ok, I will give it a try. Do you know if it is based off of Ubuntu? |
| 22:53.50 | TazzNZ | WIMPy: RTP transcoding needs "realtime" |
| 22:53.58 | portalBlock | (Wondering because of the apt repos) |
| 22:54.06 | WIMPy | That's still not realtime. |
| 22:54.07 | TazzNZ | portalBlock: it's centos |
| 22:54.14 | WIMPy | Not at all. |
| 22:54.23 | portalBlock | Meh, CentOS. |
| 22:54.46 | TazzNZ | not in the scheduling sense, but you can't really delay the packets |
| 22:54.46 | portalBlock | Wait does that even have apt? |
| 22:54.54 | TazzNZ | portalBlock: no, yum |
| 22:54.57 | TazzNZ | and rpm |
| 22:55.16 | portalBlock | I will just use this dell laptop to try it out... |
| 22:55.21 | TazzNZ | Ubuntu/Debian is apt and .deb's |
| 22:55.31 | portalBlock | Since I dont need a PCI card now :D |
| 22:55.35 | WIMPy | VOIP packets are delayed all the time. A few µs from some virtualization shouldn't make any difference. |
| 22:55.59 | TazzNZ | sure - a few won't kill it |
| 22:56.21 | WIMPy | The jitter on the network will definitely be more. |
| 22:56.22 | portalBlock | HUmm this dell has a password... |
| 22:56.28 | WIMPy | Probably even on a LAN. |
| 22:56.57 | TazzNZ | I would agree if there was only 1 or 2 VM's running on the host |
| 23:02.50 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 23:07.52 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 23:12.02 | portalBlock | What version of asterisk should I use? 11 or 1.8? |
| 23:12.16 | WIMPy | 11 or 12 |
| 23:12.22 | portalBlock | kk |
| 23:13.28 | portalBlock | The only reason I am using FreePBX is because most dialplans etc are already setup out of the box, otherwise I would just install with Ubuntu server |
| 23:13.59 | WIMPy | Are you sure that helps? |
| 23:14.05 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:14.13 | portalBlock | Nope, but it saves a bit of time. |
| 23:14.33 | WIMPy | Maybe |
| 23:14.47 | WIMPy | It could just as easily cost a lot more time. |
| 23:16.07 | portalBlock | Yeah but for basic use its good. |
| 23:16.37 | WIMPy | If you're sure thet you fall 100% within it's definition of basic usage. |
| 23:16.57 | portalBlock | Yeah I am sure |
| 23:17.58 | TazzNZ | portalBlock: and another thing - if you *every* going to hand over this system to anyone, I would suggest a web interface of some sorts |
| 23:18.29 | portalBlock | TazzNZ, hence FreePBX also |
| 23:19.05 | TazzNZ | yeah - that is what we did - the guy that took over from me is now a dCAP :D |
| 23:19.27 | portalBlock | dCAP? |
| 23:19.39 | TazzNZ | Asterisk certified pro |
| 23:20.03 | TazzNZ | (in a nutshell) |
| 23:20.21 | mbowie | He's in a nutshell? That sounds restrictive. |
| 23:20.34 | mbowie | And dark. |
| 23:20.35 | TazzNZ | meh - we prefer it that way :) |
| 23:20.43 | portalBlock | Lol |
| 23:20.50 | TazzNZ | otherwise he comes up with new ideas and all that new age stuff |
| 23:20.54 | mbowie | Ew |
| 23:21.21 | portalBlock | FreePBX is not finding my flash drive o-o (Dont laugh at me for using a flash drive :P) |
| 23:21.52 | portalBlock | its NTFS is that going to cause problems? |
| 23:22.13 | WIMPy | yes |
| 23:22.19 | Chainsaw | You can't write to NTFS in Linux. |
| 23:22.25 | Chainsaw | So yes, that may be restrictive. |
| 23:22.28 | portalBlock | Ahh |
| 23:22.30 | WIMPy | Use a real *x filesystem. |
| 23:22.30 | TazzNZ | eerrr |
| 23:22.47 | TazzNZ | that was fixed with the new fuse driver wasn't it ? |
| 23:22.57 | WIMPy | Off course you can write NTFS. But it doesn't work as system FS. |
| 23:23.04 | WIMPy | "new" |
| 23:23.17 | TazzNZ | yeah..."new" :) |
| 23:23.19 | Chainsaw | No, all of the NTFS filesystem authors concerns were overridden with a "but it worked on these 3 files I tried" FUSE driver that doesn't perform very well. |
| 23:23.21 | WIMPy | That FUSE thing has been working for years. |
| 23:23.22 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 23:23.32 | TazzNZ | it's been that long since I touch NTFS under linux |
| 23:23.45 | Chainsaw | I wouldn't trust it for any data you really care about. |
| 23:24.08 | portalBlock | What file system should I use? |
| 23:24.12 | Chainsaw | ext4 |
| 23:24.13 | WIMPy | It has been ggod enough to read/write the odd NTFS formatted pen drive. |
| 23:24.15 | portalBlock | k |
| 23:24.44 | TazzNZ | anyways - portalBlock - if you want to move data between a Windows box and a Linux box (using a usb prn drive), use FAT32 |
| 23:25.00 | Chainsaw | Unless the files are over 4GB in size. |
| 23:25.04 | TazzNZ | or be aware of the performance issues with the ntfs driver |
| 23:25.07 | TazzNZ | and that |
| 23:25.19 | WIMPy | If FAT didn't have those restrictions that sometime just make it impossible. |
| 23:25.49 | WIMPy | Windows is nice with that. It kept telling me the volume was full. |
| 23:27.28 | Chainsaw | WIMPy: FAT32 has the file size limit. FAT16 is worse, in that you can only have ~512 files in the root directory. |
| 23:27.49 | WIMPy | that's not something that limits me. |
| 23:28.15 | Chainsaw | WIMPy: Until it does, and you don't expect it so it'll take you a long time to figure out. |
| 23:28.23 | WIMPy | But the fact that 16G free space wasn't enough to write a 8G file does. |
| 23:28.37 | TazzNZ | hhhmmm wikipedia doesn't agree with Chainsaw |
| 23:28.47 | WIMPy | Can happen with ext* as well. |
| 23:28.52 | WIMPy | And the error message is equally useless. |
| 23:28.57 | TazzNZ | whoops - nevermind - misread |
| 23:29.01 | Chainsaw | WIMPy: inode limits, yes. |
| 23:29.13 | Chainsaw | TazzNZ: Citation needed. |
| 23:29.22 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 23:29.26 | file | moo |
| 23:29.37 | TazzNZ | file: s/m/p ? |
| 23:29.40 | TazzNZ | :D |
| 23:29.53 | TazzNZ | lol - that looks like I asked for something way different |
| 23:32.01 | TazzNZ | Chainsaw: I was looking for the FAT filesystem that supported files bigger than 4gig, when I saw this: |
| 23:32.09 | TazzNZ | FAT32: 2 TiB |
| 23:32.18 | TazzNZ | but that is the volume size, not the file size |
| 23:33.03 | Chainsaw | TazzNZ: The 4 gigabyte file limit in FAT32 is well documented. Particularly now that a Game of Thrones episode is about five gigs on average. |
| 23:33.36 | TazzNZ | HAHA - yeah - that is what I remember too - but 274,877,906,943 bytes (256 GiB - 1) only with FAT32+ |
| 23:33.44 | TazzNZ | that is the file size limit |
| 23:34.09 | TazzNZ | (that must be raw 1080p thou) |
| 23:34.38 | Chainsaw | Not sure that FAT32+ is well recognised. I know exFAT isn't. |