| 00:07.13 | *** join/#asterisk danfromuk (~IceChat77@2.27.39.66) |
| 00:12.15 | danfromuk | Hi. I was wondering if anyones managed to set up Asterisk to handle around 700 active calls, and around 1,000 sip peers. |
| 00:12.53 | danfromuk | Specifically if they used anything other than one asterisk box. |
| 00:13.25 | WIMPy | With more than one it's more tricky |
| 00:13.35 | youjelly | danfromuk: what kind of a dialplan is it going to run |
| 00:13.36 | WIMPy | , off course. |
| 00:13.37 | jpsharp | I would defintely use multiple boxes, if not for anything but redundancy. |
| 00:14.02 | youjelly | I had around 4 boxes, but we were using SS7 |
| 00:14.15 | danfromuk | apart from redundancy, is there any other reason to have multiple boxes? |
| 00:14.17 | WIMPy | Redundancy is a good ide, but another topic. |
| 00:14.18 | youjelly | so load balancing was being handled by the telco |
| 00:14.44 | youjelly | but it did have failover, another 4 boxes, but its more or less protocol specific |
| 00:15.06 | jpsharp | danfromuk: Depends on what you're doing it with. It you're simply passing phone calls around, you could scale up to 700 calls with enough horsepower in one box. |
| 00:15.09 | youjelly | for SIP you'll need a proxy |
| 00:15.13 | danfromuk | Its a purely SIP based solution, with no physical pstn lines. |
| 00:15.44 | youjelly | and yeah if you're just handling calls, you can do it using just one box |
| 00:15.45 | danfromuk | The sip peers make and receive calls, sometimes through a SIP Wholesale provider, and sometimes internally to other peers. |
| 00:16.09 | youjelly | we were running an extensive dial plan, an IVR |
| 00:16.09 | jpsharp | The only thing that will eat you alive with just SIP peers is if you have to do any kind of codec translation. |
| 00:16.20 | youjelly | and doing mp3 decoding and stuff |
| 00:16.22 | WIMPy | You know, it's not the number of calls that me the difference, but what you do with them. |
| 00:16.44 | WIMPy | Or lots of recordings. |
| 00:16.45 | danfromuk | Would you still load balance? Or is it just a case of having a standby box with Openser manage the switch over if the main asterisk box fails? |
| 00:17.01 | danfromuk | WIMPy: only a handful of calls are recorded at the moment. |
| 00:17.39 | jpsharp | If you want to expand that number of recordings you'll need either stupid fast I/O or another box. |
| 00:17.49 | danfromuk | True. |
| 00:18.06 | danfromuk | I was just trying to work out how many calls I could manage with the current solution. |
| 00:18.15 | youjelly | HDDs support around 40 MB/s of write |
| 00:18.17 | danfromuk | If there is no recording. |
| 00:18.37 | youjelly | a call is 8kBytes/sec |
| 00:18.55 | WIMPy | per direction |
| 00:19.28 | danfromuk | Do people generally load balance in this situation? Or do they have a backup server ready in order to provide redundancy? |
| 00:20.00 | WIMPy | Are people generally chepskates or not? |
| 00:20.39 | WIMPy | it's up to ypu how much availability/reliability you need/want. |
| 00:20.50 | danfromuk | WIMPy: either way, its still 2 or 3 servers running. |
| 00:20.59 | jpsharp | I design my systems so that they load balance, but either machine is sized so it can take on the full load if needed. |
| 00:21.00 | danfromuk | its just whether they are used or not. |
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| 00:22.11 | jpsharp | Keeps the kinks worked out of both systems at the same time. Nothing sucks more than failing over to your standby system only to realize that it has an old config or your config sync system doesn't work right. |
| 00:22.50 | jpsharp | Or your failover system in general gives you finger when it tries to fail over. |
| 00:23.09 | youjelly | lol |
| 00:25.03 | danfromuk | thats a good point. |
| 00:25.10 | danfromuk | ok, thanks. something to think about. |
| 00:25.17 | jpsharp | Not that I would know anything about that. |
| 00:25.27 | danfromuk | does everyone use openser to manage load balancing? |
| 00:25.52 | jpsharp | Tis what I've used. |
| 00:26.42 | danfromuk | ok great. thanks for the advice. |
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| 01:11.48 | endiruna | hi i would like for my mom to have a ip phone to connect to my server so that i can call her and she me. is that possible? |
| 01:12.07 | jpsharp | Yes. |
| 01:12.39 | jpsharp | That's the short answer. The long answer depends on where your server is. |
| 01:14.13 | slav3_kitten | qubec! |
| 01:14.38 | WIMPy | And what colour? |
| 01:14.40 | jpsharp | Then you have to pay customs duties when your packets cross the border. |
| 01:14.54 | slav3_kitten | and they have to be in french format |
| 01:15.36 | jpsharp | Le Pack'et |
| 01:16.50 | Happy0 | lmao |
| 01:19.15 | endiruna | are there some tutorials. never used asterisk |
| 01:19.35 | jpsharp | ~book |
| 01:19.36 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:19.42 | jpsharp | There you go. |
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| 01:29.37 | endiruna | jpsharp i was looking more for a tutorial. the whole manual seems kind of overwhelming :D |
| 01:29.46 | endiruna | but thanks |
| 01:30.30 | ectospasm | <PROTECTED> |
| 01:36.28 | jpsharp | Yep. Its pretty much "Asterisk for dummies". |
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| 02:00.48 | TomCat2 | so can anyone elaborate on what 'phishing' has to do with someone being able to hack someone else's asterisk box? |
| 02:02.07 | TomCat2 | o_O |
| 02:04.11 | TomCat2 | nvm I'm being facetious |
| 02:09.41 | ChannelZ | uhm.. |
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| 02:26.39 | endiruna | how do i get called if i do ot have a public ip? |
| 02:27.02 | endiruna | i guess i am missing the keyword to look for. |
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| 03:09.05 | pyther | Can anyone point me in the right direction to setup call waiting? |
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| 03:30.03 | carrar | google |
| 03:30.26 | TomCat2 | anyone know off hand how to make my PAP2, which is behind a NAT/Router, but that I have port 5061 mapped to pass through, to register with my outside Asterisk server using the external IP? Currently it's registering, but with a 192.168.*.* IP |
| 03:30.41 | TomCat2 | my external IP is static |
| 03:33.31 | TomCat2 | something besides hard-wiring it into my sip.conf |
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| 04:21.51 | techhelper1 | well, i want to use the ilbc codec, i installed asterisk from apt-get in ubuntu |
| 04:22.02 | techhelper1 | so how do i compile and install this |
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| 04:38.40 | techhelper1 | nevermind i figured it out |
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| 08:16.54 | johnf | Has anyone managed to get wanpipe compiled with ubuntu 12.04. I'm getting "wanpipe: no symbol version for dahdi_hdlc_putbuf" on modprobe wanpipe |
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| 08:25.34 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 08:26.36 | verywiseman | can i write asterisk variable to a file? |
| 08:40.58 | *** part/#asterisk johnf (~johnf@124-168-183-82.dyn.iinet.net.au) |
| 08:56.03 | TomCat2 | yes |
| 09:31.48 | verywiseman | i write some commands after Dial app , these commands don't execute after hang up , why? Note i use Dial app with 'g' option |
| 09:45.22 | ChannelZ | g only makes the dialplan continue when the called channel hangs up |
| 09:45.47 | ChannelZ | Is that the end hanging up? |
| 09:53.24 | verywiseman | ChannelZ, pls ,i know , |
| 09:53.55 | verywiseman | ChannelZ, i know , but when call hang up , the following commands didn't execute. |
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| 11:04.46 | elliot98 | enters and takes a seat |
| 11:05.03 | WIMPy | Tickets please! |
| 11:05.36 | elliot98 | throws out an INVITE packet |
| 11:06.02 | WIMPy | There's no guest list today. |
| 11:06.59 | elliot98 | checks over sip.conf |
| 11:07.47 | elliot98 | gets up and goes to the default context |
| 11:08.16 | elliot98 | tcpdumps show incoming INVITEs, but doesn't look like Asterisk is handling them |
| 11:08.34 | elliot98 | this happens once is a while...what gives? |
| 11:08.57 | WIMPy | Did you bind to 0.0.0.0 or something specific? |
| 11:13.31 | elliot98 | it usually works, but sometimes it's as if Asterisk ignores some packets every so often |
| 11:14.18 | WIMPy | Did you try to find them with sip debug? |
| 11:14.32 | elliot98 | tcpdump |
| 11:17.42 | elliot98 | WIMPy: are there any sip debug parses or ways to through the captures into a pcap file? |
| 11:18.46 | WIMPy | Not that I know. |
| 11:20.02 | WIMPy | thinks that chan_sip needs a real debugging option instead of just dumping packets. |
| 11:21.27 | elliot98 | chan_sip could invoke libcap |
| 11:22.28 | WIMPy | It could, but usually using tcpdump (or whatever) should be justas good. |
| 11:28.33 | elliot98 | the only thing is if there is something up the Asterisk that's it's ignoring packets |
| 11:29.44 | elliot98 | can Asterisk use RTP instead of UDPTL for T38? |
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| 13:27.13 | pyther | Is there a tool that'll let me make my own voice prompts using a computurized voice? |
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| 13:45.12 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 13:45.15 | s[x] | hey all |
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| 14:50.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:59.57 | endiruna | what is a good sip protocol i could use for a not very speedy connection |
| 15:00.15 | endiruna | how is rfc 3261 |
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| 15:45.13 | pyther | Is it possible to check if a user is registered and route calls based on this information? |
| 15:45.39 | pyther | Basically when my andoird phone is registerd I want to follow a different procedure than when its not registered |
| 15:46.01 | WIMPy | See CHANISAVAIL or try and check DIALSTATUS. |
| 15:46.35 | WIMPy | Using qualify may help as well. |
| 15:48.03 | sekil | pyther: that should be easy.. |
| 15:48.20 | pyther | great, any good example snippets |
| 15:48.49 | pyther | basically the way I have it right now it rings my internal phones, then plays a message, and then uses Follow Me to ring my cell and then go to voice mail |
| 15:49.06 | pyther | I want to trying the cell if the android is registered |
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| 16:06.44 | pyther | nexten => 704,1,ChanIsAvail(SIP/android) is that the right syntax? |
| 16:06.54 | pyther | it seems that AVAILSTATUS is always 0 |
| 16:24.28 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:43.09 | *** join/#asterisk nutxase (~nutxase@41-133-161-25.dsl.mweb.co.za) |
| 16:43.27 | nutxase | anyone see dahdi dropping inbound calls? |
| 16:43.47 | WIMPy | When? Why? |
| 16:44.02 | WIMPy | What interface? |
| 16:44.39 | nutxase | isdn either just as it hits the ivr it gets hangup cause 27 or whilst you are speaking |
| 16:44.41 | nutxase | not all calls |
| 16:45.36 | nutxase | tried swopping the pstn card to a different pci slot today |
| 16:46.19 | WIMPy | Where do you get the cause? From the caller? |
| 16:46.27 | nutxase | no from the asterisk cli |
| 16:46.40 | WIMPy | For an incomming call? |
| 16:46.47 | nutxase | yes |
| 16:47.02 | WIMPy | That doesn't make sense. |
| 16:47.05 | WIMPy | Any more information? |
| 16:47.39 | nutxase | all it says is hangup cause 27 and then the caller will just hear ringing in their side and not the ivr playing |
| 16:47.50 | nutxase | but if the call cuts whilst they are speaking then it goes dead |
| 16:48.05 | nutxase | so asterisk picks the call up then dahdi kills it |
| 16:48.15 | WIMPy | That doesn't make sense, either. |
| 16:48.35 | WIMPy | doesn't understand the last sentence. |
| 16:48.43 | nutxase | so asterisk answers the call |
| 16:48.58 | nutxase | then suddenly u get hangup cause 27 and the call does not ring into the system anymore |
| 16:49.27 | WIMPy | There is something very wrong, it seems. |
| 16:49.45 | nutxase | yes and have no idea what it is or where to look anymore |
| 16:50.00 | WIMPy | enable pri debug and see if you can pb an occasion. |
| 16:50.11 | nutxase | ok 2 secs |
| 16:50.47 | WIMPy | debug 2 might tell even more. |
| 16:50.59 | nutxase | debug2? |
| 16:51.13 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 16:51.16 | WIMPy | pri set debug 2 span X. |
| 16:52.16 | timahvo1 | Hi everybody , just did a factory reset on a cisco 7942 IP phone but the phone doesn't pick up an IP address after that. anyone experience a similar problem ? any pointers ? |
| 16:53.52 | nutxase | http://pastebin.com/6g0riU0L |
| 16:55.29 | WIMPy | Do you have multiple spans? |
| 16:55.36 | WIMPy | I don;t see a call there. |
| 16:55.46 | nutxase | yes 4 |
| 16:55.55 | nutxase | see the call right at the top where it enters custom/afterhours |
| 16:56.01 | WIMPy | Oh right. It says so on every line... |
| 16:56.19 | pyther | Would someone mind taking a look at my dial plan letting me know if there are better ways to do things? http://pastebin.com/VxihCrA0 |
| 16:56.32 | WIMPy | Then enable pri debug for all if you don;t know where a call might come in. |
| 16:56.59 | nutxase | i have done so it happends exactly the same on all spans :/ |
| 16:57.19 | WIMPy | But I didn't see a call coming in there. |
| 16:57.39 | WIMPy | But you seem to have a layer2 issue, I think. |
| 16:58.15 | nutxase | how do i fix a layer 2 issue? |
| 16:58.21 | nutxase | right at the top u can see the call coming in |
| 16:58.30 | nutxase | <PROTECTED> |
| 16:58.38 | WIMPy | Give me a full call to see what's really going on. |
| 16:58.44 | nutxase | ok 1 sec |
| 16:59.04 | WIMPy | That's after it came in on dahdi. |
| 16:59.40 | WIMPy | Can you call out? |
| 17:00.34 | nutxase | yes out is perfect |
| 17:03.00 | WIMPy | And BTW: Why are incomming calls hitting s? |
| 17:03.21 | WIMPy | That doesn't sound right, either. |
| 17:03.31 | nutxase | thats how freepbx sets it :/ |
| 17:04.05 | nutxase | what should it hit? |
| 17:04.19 | WIMPy | The called number. |
| 17:04.40 | WIMPy | s shouldn't be possible in TE mode. |
| 17:04.47 | nutxase | but its not a did its 1 of 4 isdn in a hunt group if i enable the did the call fails |
| 17:05.42 | WIMPy | Sounds like ptp lines with ddi? |
| 17:05.57 | nutxase | ptmp lines 4 of them yea isdn |
| 17:06.23 | WIMPy | With MSNs or with DDI? |
| 17:06.49 | nutxase | no ddi |
| 17:06.54 | nutxase | just 1 number per isdn line |
| 17:07.35 | WIMPy | Per line? Didn't you say hunt group? |
| 17:08.17 | nutxase | so there is 4 isdn lines each with a number and they are in a hunt group from the telco so if one is busy the next one rings etc |
| 17:08.19 | WIMPy | Is that DSS1 or something special? |
| 17:08.33 | WIMPy | Stange setup. |
| 17:08.44 | nutxase | normal isdn 2a |
| 17:09.09 | WIMPy | a? |
| 17:09.19 | nutxase | or isdn2 |
| 17:09.57 | WIMPy | isdn ha;s multilink capabilities, so making single links and put them in a hunt group somewhere else is not what you'd normally do. |
| 17:10.23 | WIMPy | Anyway. Do you have a full call? |
| 17:10.47 | nutxase | just trying to reproduce one now |
| 17:10.51 | nutxase | its not all calls |
| 17:11.11 | WIMPy | (and are you sure your telco kows what they're doing?) |
| 17:11.30 | nutxase | they say so lol |
| 17:14.29 | nutxase | also wats weird is if i change in chan_dahdi.conf to go to a diff context it still goes to from-pstn |
| 17:15.24 | WIMPy | the context must be before the channel => line. |
| 17:16.00 | nutxase | looks like [channels] |
| 17:16.01 | nutxase | context=incoming |
| 17:16.20 | nutxase | dont have a channel= line |
| 17:16.47 | WIMPy | Then you don't have channels. |
| 17:16.48 | nutxase | let me show u my chan_dahdi file |
| 17:16.53 | WIMPy | Look for included files. |
| 17:18.33 | nutxase | found it but still wants to go to s :( |
| 17:19.33 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:22.17 | nutxase | maybe i must add a custom dialplan and add an answer and a silence/1 |
| 17:22.48 | WIMPy | To do what? |
| 17:22.56 | nutxase | nevermind same error |
| 17:23.52 | WIMPy | That error has nothign to do with your dialplan. |
| 17:32.41 | nutxase | yea |
| 17:32.46 | nutxase | got the error hold on |
| 17:33.28 | *** join/#asterisk Bogus8 (~Nunya@ip184-190-200-28.lf.br.cox.net) |
| 17:35.11 | Bogus8 | What would be some decent basic VOIP phones that I could pick up cheap that would allow me to test and then get put into production later. Doesn't need to be super fancy but multiple lines would be a plus. Even refurb or used would be ok with me. I'm mainly looking for CHEAP on these first two phones. |
| 17:35.58 | nutxase | yealink |
| 17:36.50 | nutxase | doesnt fit into pastebin can i email u the txt doc? |
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| 17:38.07 | WIMPy | Bogus8: The cheapest is a piece of free software. |
| 17:38.43 | WIMPy | nutxase: Wow. Do you have anywhere else you can put it? How big is it? |
| 17:39.05 | Bogus8 | WIMPy: hmm, true, I could use a pc as a VOIP phone for testing purposes and be confident that all this would work when a "real" phone was plugged into the network since they are just miniture computers doing what that software does as well. |
| 17:39.27 | WIMPy | exactely |
| 17:39.37 | nutxase | 1meg |
| 17:40.51 | Bogus8 | For shits and giggles what about the polycom 501's? |
| 17:41.19 | WIMPy | NFI. People in here seem to love Polycom. |
| 17:42.03 | Bogus8 | I could pick up a few of them for about 20 bucks each. |
| 17:42.10 | Bogus8 | used of course. |
| 17:45.54 | Bogus8 | While I'm here... is google voice working with asterisk these days? I heard there was a lot of work arounds needed recently. |
| 17:47.09 | Bogus8 | I'd like to find a good guide on how to go from knowing nothing about VOIP to being up and running with google voice and some SIP phones. I have several linux servers already in production and I'm comfortable with the ins and outs of linux... I just know next to nothing about VOIP, asterisk and google voice. |
| 17:48.36 | WIMPy | Only Google knows what Google is going to do. |
| 17:51.09 | carrar | but whatever they do, it will involve tracking you |
| 17:53.55 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:55.53 | Bogus8 | carrar: I hear ya! Sad as they were the only ones that didn't give up your search queries to the government without a court order |
| 17:57.18 | ChannelZ | I need to finish my primer. |
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| 18:32.00 | pyther | Would appreciate it if someone could look over my dial plan. Let me know if it looks good or if there are better ways to accomplish what I'm tyring to do. http://pastebin.com/VxihCrA0 |
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| 21:34.50 | *** join/#asterisk brokedcomputer (~ce@184.71.199.202) |
| 21:34.58 | brokedcomputer | Hey Guys, how goes it? |
| 21:35.28 | pyther | well, I finally have a working asterisk setup :) |
| 21:35.47 | brokedcomputer | Excellent! I just came online for when I need to set it up on the Ubuntu server I am setting up right now. |
| 21:36.07 | brokedcomputer | Gonna do a Samba server/Asterisk Machine |
| 21:36.26 | pyther | cool, much linux experience? |
| 21:36.43 | brokedcomputer | Enough that I know how to use commandline better than the GUI? |
| 21:37.08 | pyther | thats a good start, the command line is powerful |
| 21:37.12 | brokedcomputer | :) |
| 21:37.32 | brokedcomputer | I have two servers that I run that are Ubuntu and dedicated one to Apache, and one to Minecraft/fileserver |
| 21:39.28 | brokedcomputer | Should I stick with Ubuntu 10.04 for this, or is there a stable release of Asterisk for Ubuntu 12? |
| 21:40.26 | jpsharp | Install an OS, build Asterisk from scratch. |
| 21:40.35 | jpsharp | Packages will cause you heartburn. |
| 21:40.37 | brokedcomputer | I can do that. :P |
| 21:40.49 | brokedcomputer | Where can I get a list of requirements for building it? |
| 21:40.54 | pyther | jpsharp: really? |
| 21:41.05 | brokedcomputer | ? |
| 21:41.35 | jpsharp | The requirements depend on what you want to do. A basic PBX doesn't require external libraries. |
| 21:41.45 | brokedcomputer | Heh, I would like to use AsteriskNOW with PBX 2.02 |
| 21:41.52 | brokedcomputer | 2.0.2 * |
| 21:41.54 | jpsharp | pyther: The packages aren't being kept up to date as the should be. |
| 21:42.17 | brokedcomputer | I am going to have three phones with 3 incoming lines, |
| 21:42.25 | brokedcomputer | and all IP phones |
| 21:42.36 | pyther | so my softphone tells me I have a voice messages (yay) but when I click on the notification to listen I get 404 |
| 21:42.36 | brokedcomputer | We have two to test with and one analoug handset. |
| 21:42.39 | jpsharp | incoming lines with hardware? |
| 21:42.46 | pyther | Call from 'android' (76.18.191.214:44206) to extension 'null' rejected because extension not found in context 'from-internal'. |
| 21:42.46 | brokedcomputer | Yeah |
| 21:43.00 | brokedcomputer | The location of the system is not convenient for soft lines |
| 21:43.11 | pyther | is that a soft phone issue or is there a setting you can set in asteriks to specify the voicemail extension? |
| 21:43.29 | jpsharp | That's a softphone issue. |
| 21:43.43 | jpsharp | Unless you want to create an extesion called "null" |
| 21:44.08 | pyther | can create an extension called null? |
| 21:44.14 | jpsharp | brokedcomputer: You'll want to install from source, then. It seems the package builders can't get DAHDI right. |
| 21:44.16 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 21:44.25 | brokedcomputer | For reference its a Xeon dual processor with 4 cores each at 2.5Ghz, with 8GB ram so far. |
| 21:44.34 | jpsharp | You can create an extension named anything. |
| 21:44.35 | brokedcomputer | *Second processor is not installed at the moment. |
| 21:44.52 | jpsharp | exten => xyzzy,1,Playback("yaaaaaaay!") |
| 21:45.15 | jpsharp | ohwow. That's serious overkill for that many lines. :) |
| 21:45.18 | pyther | really sweet! |
| 21:45.21 | brokedcomputer | Hahaha, yeah |
| 21:45.31 | brokedcomputer | But its also going to be doing quite a few other things :P |
| 21:47.30 | pyther | jpsharp: would you mind looking over my dial plan? http://pastebin.com/CZwCqRK7 |
| 21:48.33 | jpsharp | I don't see anything glaringly wrong with it. |
| 21:48.49 | pyther | sweet, just wanted make sure I wasn't doing anything stupid |
| 21:48.58 | jpsharp | brokedcomputer: Asterisk is pretty lightweight unless you're doing codec translation or call recording. |
| 21:51.00 | jpsharp | I run Asterisk for the house (VOIP only), Apache, Samba, a media server, 2 instances of Minecraft, a WRF weather modelling system, and a weather datastream recorder on a quad 1.866Ghz Xeon with 8GB of RAM. |
| 21:51.01 | brokedcomputer | Thats fine, |
| 21:51.43 | brokedcomputer | The customer built the machine and then asked me to use it, I explained that its Overpowered for anything linux, but its what they wanted. |
| 21:52.08 | pyther | brokedcomputer: lies, nothing is over powered for linux |
| 21:52.20 | brokedcomputer | Hahahaha, |
| 21:52.37 | brokedcomputer | Also true. :P What I meant was, for their purpose. |
| 21:52.58 | *** join/#asterisk lorsungcu (~lorsungcu@65.103.31.36) |
| 21:54.09 | lorsungcu | anyone have experience using ODBC_FETCH with multirow results? |
| 21:57.22 | brokedcomputer | Ok, |
| 21:57.39 | brokedcomputer | hahaha, so where should I start now that there is a dissagreement on the packages part? |
| 22:00.47 | jpsharp | Are you going to use hardware cards installed into the server for the inbound lines? |
| 22:01.08 | brokedcomputer | Yes |
| 22:02.37 | jpsharp | Then install the kernel sources (packages are fine for that). Download and install DAHDI from asterisk.org. Download and install Asterisk. When you ./confgiure asterisk, it will detect that you've installed DAHDI and build the appropriate channels. |
| 22:03.11 | jpsharp | DAHDI being the drivers/driver subsystem for the cards. |
| 22:03.20 | jpsharp | And then you're done & ready to configure. |
| 22:04.37 | brokedcomputer | Interesting |
| 22:06.21 | brokedcomputer | Ok, I am going to do updates first, then we will start on the DAHDI |
| 22:06.35 | jpsharp | And that way, later on when you want to add something like Fax support or something else, you're not hamstrung by the package. |
| 22:06.55 | brokedcomputer | I am not sure what you mean by that, |
| 22:07.11 | brokedcomputer | Can I send Fax rewquests to the server as well through IP? |
| 22:07.19 | brokedcomputer | requests* |
| 22:09.15 | jpsharp | Asterisk can act like a fax machine. Send it a TIFF file and a phone number and it'll send it out. Set it up to receive and it can answer a fax call, turn it into a TIFF file, and do something with it. |
| 22:09.31 | brokedcomputer | EPIC! |
| 22:09.36 | brokedcomputer | I did not know that. |
| 22:09.46 | jpsharp | That's the simplistic description, though. It takes some dialplan work to make it do that, but it can be done. |
| 22:10.15 | brokedcomputer | You don't learn how to fix a car till you know you need to know :) |
| 22:10.55 | brokedcomputer | Now this is a Digium card, specific to Asterisk, does that change the need for the DAHDI? |
| 22:11.05 | jpsharp | That says you need DAHDI. |
| 22:11.18 | jpsharp | DAHDI is the driver system for the Digium cards. |
| 22:11.24 | brokedcomputer | OH hahahah :p |
| 22:12.00 | brokedcomputer | Just updated the kernel to 2.6.3, should be ready in a moment or two now. |
| 22:12.11 | jpsharp | Make sure you install the kernel source package. |
| 22:12.25 | brokedcomputer | Not sure what you mean? |
| 22:12.59 | pyther | is there anything I want to enable/disable in the build option selection? |
| 22:13.37 | jpsharp | There's the kernel package, that just has the kernel and modules. Then there's the kernel source package which has the source tree and header files needed to build drivers, which is what DAHDI will want to do. |
| 22:15.31 | brokedcomputer | What is the name for the Kernel source packages? |
| 22:17.14 | brokedcomputer | Hehe never mind figured it out |
| 22:17.15 | brokedcomputer | :P |
| 22:17.19 | brokedcomputer | Linux-source |
| 22:19.45 | jpsharp | Make sure it is the right one for the kernel you updated too :) |
| 22:19.51 | jpsharp | Not that I've ever done that. |
| 22:21.59 | brokedcomputer | hahaha, 2.6.32 is the latest and the one. :P |
| 22:22.20 | brokedcomputer | awww there's no DAHDI package? :P |
| 22:30.48 | brokedcomputer | ... how do I compile the dahdi source? I am in the folder... |
| 22:39.08 | *** join/#asterisk MariusKarthaus (~quassel@541F8611.cm-5-8c.dynamic.ziggo.nl) |
| 22:43.06 | brokedcomputer | Ok, DAHDI is installed, getting Asterisk |
| 22:43.10 | brokedcomputer | 1.8.11 |
| 22:45.29 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 22:45.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:56.34 | brokedcomputer | Asterisk is Making now. |
| 23:00.04 | brokedcomputer | Is there a web interface for Asterisk when built from scratch? |
| 23:04.50 | brokedcomputer | jpsharp you still there? Its ready and running |
| 23:15.22 | WIMPy | No |
| 23:15.31 | brokedcomputer | Apparently :P |
| 23:15.39 | WIMPy | And you don't really want it anyway. |
| 23:15.55 | brokedcomputer | So do you know why the http.conf file is not in etc/asterisk/? |
| 23:17.46 | WIMPy | The webserver is for the http manager (no idea if that's useful for anythign) and probably also for provisioning (haven't tried that from withinn Asterisk). |
| 23:18.19 | brokedcomputer | Where would I find the conf files if the're not in etc/asterisk? |
| 23:18.35 | lorsungcu | what conf files are you looking for? |
| 23:18.47 | brokedcomputer | Asterisk's |
| 23:18.57 | lorsungcu | they will be in /etc/asterisk |
| 23:19.21 | brokedcomputer | thats what I am saying the're not in there |
| 23:19.30 | WIMPy | 'make samples' |
| 23:19.33 | *** join/#asterisk Curs0r (~Curs0r@unaffiliated/curs0r) |
| 23:19.42 | lorsungcu | from the source directory, do what wimpy says |
| 23:19.55 | WIMPy | But be aware that some of the samples are more like documentation and not meant as usable configs. |
| 23:20.03 | lorsungcu | and you' |
| 23:20.11 | lorsungcu | ll end up loading a lot of stuff you probably dont want |
| 23:20.21 | WIMPy | That as well. |
| 23:20.36 | brokedcomputer | How can I make it so that I can access the GUI before I configure anything, I want a clean conf. |
| 23:20.49 | lorsungcu | www.freepbx.org |
| 23:20.58 | WIMPy | And then change to #freepbx |
| 23:21.14 | lorsungcu | unless you installed the asterisk gui, there is no interface |
| 23:21.50 | brokedcomputer | thats what I am installing |
| 23:22.09 | brokedcomputer | or... trying, it is installed, but I need to change files that don't exsist |
| 23:22.14 | brokedcomputer | exist* |
| 23:22.26 | lorsungcu | download the asteriskNOW + gui ISO |
| 23:22.43 | lorsungcu | theres little/no reason to do that from source.. |
| 23:22.58 | brokedcomputer | ... I am installing this on Ubuntu, you cannot use the ISO to install onto Ubuntu can you? |
| 23:22.59 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 23:23.20 | lorsungcu | no, unless you've got KVM running or something |
| 23:25.46 | Curs0r | I hate to display my mental handicaps in public but can someone translate the following line to 'stupid' for me? |
| 23:25.48 | Curs0r | Jul 22 16:24:43] WARNING[1534]: chan_sip.c:3906 retrans_pkt: Maximum retries exceeded on transmission 926ae8d31b6f422f8bcf9db4e36dffa4 for seqno 8086 (Critical Response) -- See doc/sip-retransmit.txt. |
| 23:26.34 | jpsharp | It kept retransmitting a packet until it gave up. |
| 23:26.41 | Curs0r | My music on hold extension keeps doing that to me after 20 seconds consistently |
| 23:26.55 | Curs0r | Perfect every time... for 20 seconds lol |
| 23:27.09 | Curs0r | And, thank you jpsharp |
| 23:27.31 | brokedcomputer | how do I bring up the server CLI for Asterisk? |
| 23:27.38 | lorsungcu | asterisk -rvvv |
| 23:27.50 | Curs0r | sudo asterisk -r on the ubuntu box I'm playing with |
| 23:28.28 | brokedcomputer | better question, how do I quit it now that I got what I needed? |
| 23:28.38 | lorsungcu | exit |
| 23:28.42 | lorsungcu | or control-c |
| 23:29.00 | brokedcomputer | Sure it works now... I tried that after installing it and running -vvvc |
| 23:29.21 | Curs0r | So, since I am getting the music for 20 seconds before it dies each time. I'm thinking either the server or the client is properly registering that the call is connected. Now if I knew what to do about that, I'd be all set.... to the google salt mines I guess :) |
| 23:29.37 | Curs0r | is not* |
| 23:29.46 | lorsungcu | cursor, i believe that's some nat issue |
| 23:29.58 | jpsharp | -vvvvc starts it attached to the TTY you're logged into. If you quit from there, asterisk dies. |
| 23:30.19 | Curs0r | lorsungcu, possible I need to forward a port on the remote sip extension? |
| 23:30.29 | lorsungcu | i think so |
| 23:30.53 | brokedcomputer | So I quit the asterisk server when I tried to quit out of the CLI? |
| 23:30.59 | lorsungcu | i think if the remote nat port changes/gets removed, asterisk continues to send to it but doesnt get a response |
| 23:31.02 | lorsungcu | someone will correct me |
| 23:31.06 | Curs0r | That is finally straying into the area of things I can actually figure out hehe |
| 23:31.21 | jpsharp | brokedcomputer: Yes, if you started Asterisk with -c. |
| 23:31.29 | brokedcomputer | ... sheesh, |
| 23:31.39 | lorsungcu | broked |
| 23:31.42 | brokedcomputer | so is it running properly now if I started it with -rvvv? |
| 23:31.56 | lorsungcu | -r connects to a running cli |
| 23:32.00 | lorsungcu | if you are able to do that |
| 23:32.02 | brokedcomputer | Oh good |
| 23:32.02 | jpsharp | What he said. |
| 23:32.14 | lorsungcu | there is an instance of it up |
| 23:32.24 | brokedcomputer | How do I make it see the conf files I made in /etc/asterisk for the GUIU? |
| 23:32.27 | brokedcomputer | GUI* |
| 23:32.34 | lorsungcu | no idea |
| 23:32.39 | brokedcomputer | Since there were no conf files there. |
| 23:32.53 | lorsungcu | what gui are you using? |
| 23:33.03 | brokedcomputer | Asterisk-GUI |
| 23:33.16 | brokedcomputer | on port 8088 <- I checked and its not active. |
| 23:33.22 | lorsungcu | cursor, what are your sip nat settings? might want to check that those are proper, as well |
| 23:33.31 | brokedcomputer | Heh which brings me to why I was trying to talk to the CLI |
| 23:33.48 | Curs0r | lorsungcu, the sip nat settings in the /etc/asterisk/sip.conf? |
| 23:34.02 | lorsungcu | yeah |
| 23:34.02 | Curs0r | I don't even know if I set anything related to nat in that yet |
| 23:34.08 | Curs0r | Checking... |
| 23:35.09 | brokedcomputer | What command do I use to restart the process? See if that grabs the conf files. |
| 23:35.29 | lorsungcu | try /etc/init.d/asterisk restart |
| 23:36.42 | jpsharp | Or you can just "reload" from the CLI. |
| 23:36.55 | brokedcomputer | Hmmm didnt work, gave me "no such file or directory" |
| 23:37.13 | jpsharp | or "core restart now" from the CLI. |
| 23:37.33 | Curs0r | lorsungcu, [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes directmedia=no host=dynamic |
| 23:37.33 | Curs0r | <PROTECTED> |
| 23:37.52 | lorsungcu | wahts in [global], cursor? |
| 23:38.09 | lorsungcu | broked, try this? http://www.powerpbx.org/content/asterisk-gui-v2-install-centos-v5x-asterisk-v14x-asteriskgui-v2x |
| 23:38.32 | lorsungcu | its centos, but might get you pointed in the right direction. i've never used the asterisk gui, except on one of their appliances... |
| 23:39.03 | lorsungcu | or http://www.asterisk.org/asterisknow/install-related |
| 23:39.30 | Curs0r | lorsungcu, there is still so much commented default stuff in this document I think I'd better just pastebin it. It's difficult to navigate via ssh/nano :) |
| 23:39.36 | lorsungcu | go for it |
| 23:40.09 | Curs0r | Now if I can just remember that cli command to cat a file out to pastebin :) |
| 23:42.21 | brokedcomputer | Hmmm, I am getting an interesting loop effect from the GUI? |
| 23:42.36 | lorsungcu | youll need to be more specific :) |
| 23:42.38 | brokedcomputer | It loads the main page then loads the extensions.conf and does it all over again |
| 23:44.08 | brokedcomputer | 8080/asterisk/static/config/index.html <- this is what page I am at, and it just keeps reloading? |
| 23:45.05 | lorsungcu | sorry, no idea. what are you trying to do that you need the GUI on an ubuntu machine? |
| 23:45.38 | brokedcomputer | Thats so far the ONLY way I know how to set it up and I am making this for my church, I want them to be able to configure new phones through GUI later. |
| 23:45.54 | lorsungcu | do you need the ubuntu machine for something? |
| 23:46.18 | lorsungcu | freepbx will configure phones for you, etc, without ever touching a command line, if you dont want to... |
| 23:47.02 | brokedcomputer | Please don't redirect me to another system, we are using Ubuntu because this is also a Samba host, and other things. |
| 23:47.19 | brokedcomputer | I also prefer Ubuntu, and would like to use the computer for more than JUST a Phone box. |
| 23:47.56 | lorsungcu | i would not run asterisk alongside anything else |
| 23:48.15 | lorsungcu | find a small desktop and use it exclusively |
| 23:48.18 | brokedcomputer | On a quad core with 1TB harddrive space, and 4GB ram, you don't think I should run anything else? |
| 23:48.23 | lorsungcu | correct. |
| 23:48.32 | brokedcomputer | Why> |
| 23:48.45 | lorsungcu | security and timing, mostly |
| 23:48.54 | Curs0r | Hrm, either it's links2 or xming but the pastebin form isn't submitting hehe |
| 23:49.12 | lorsungcu | the more services you have running, the easier getting into it from outside will be |
| 23:49.14 | brokedcomputer | Its got hard lines coming into it, its not going to be on the net directly, there will be no outside access to it through the asterisk side. |
| 23:50.08 | lorsungcu | doesn't change my opinion, sorry |
| 23:51.52 | brokedcomputer | I just want to configure the system, are you able to help? |
| 23:54.22 | lorsungcu | like I said, i've got no experience with asterisk gui; if you want to go that route, get freepbx and put it on a separate machine. |
| 23:54.43 | lorsungcu | maybe someone else here knows more about it than me, though |
| 23:55.27 | brokedcomputer | Thats not what I stated, I would like help configuring, it doesnt have to be with the GUI, step by step is the best place. |