| 00:12.08 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:14.22 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 00:15.13 | rotten777 | wth... |
| 00:15.23 | rotten777 | flowroute won't register sip :X |
| 00:16.26 | p3nguin | register => userid:password@sip.flowroute.com |
| 00:16.48 | rotten777 | yeah i've got that |
| 00:16.56 | rotten777 | their site actually generates the conf content for you |
| 00:20.02 | rotten777 | what does insecure do ? |
| 00:20.13 | rotten777 | insecure=port,invite |
| 00:20.16 | rotten777 | does that work behind nat? |
| 00:22.08 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 00:22.30 | p3nguin | port,invite tells asterisk to ignore the port that the request came from and don't require authentication on the initial invite. Yes it works behind NAT. |
| 00:25.33 | rotten777 | does this look good to you? |
| 00:25.33 | rotten777 | http://pastebin.com/fghBJ8my |
| 00:25.40 | rotten777 | I'm about ready to give up |
| 00:25.44 | rotten777 | i dont know what is going on |
| 00:27.50 | p3nguin | Does it register? |
| 00:27.55 | rotten777 | nope |
| 00:28.03 | rotten777 | request sent |
| 00:28.07 | rotten777 | thats it |
| 00:29.26 | WIMPy | insecure=port will obviousely not work with NAT. That requires explicit port forwarding. |
| 00:30.14 | p3nguin | Maybe it doesn't do any good with NAT, but using it does not make it not work. |
| 00:30.32 | rotten777 | what ports do you want me to forward? 5060 already is on udp |
| 00:30.33 | WIMPy | That's correct. |
| 00:30.45 | p3nguin | UDP 5060 and the UDP range in rtp.conf |
| 00:30.50 | WIMPy | That's the one. |
| 00:31.29 | WIMPy | You could try to use TCP to avoid NAT and insecure issues. |
| 00:31.40 | WIMPy | If it's supported... |
| 00:31.43 | p3nguin | The providers almost never support it. |
| 00:32.37 | anonymouz666 | 300 seats from 1.4 to 1.8 going on RIGHT NOW |
| 00:32.41 | anonymouz666 | :) |
| 00:33.40 | adeel | this would have come in handy the otherday for a few people who were trying to manipulate sensord data....http://wiki.bash-hackers.org/syntax/pe#substring_expansion |
| 00:33.53 | adeel | whoops, wrong channel |
| 00:33.58 | rotten777 | i'm 99% sure the ports are forwarded |
| 00:34.35 | p3nguin | What kind of router do you have? |
| 00:34.41 | p3nguin | I hope it's not a Belkin. |
| 00:34.43 | rotten777 | mikrotik rb750g |
| 00:34.45 | rotten777 | lol no way |
| 00:34.55 | rotten777 | routing i'm good at |
| 00:35.01 | rotten777 | i use routerboards |
| 00:35.06 | rotten777 | voip... not so much |
| 00:38.02 | rotten777 | just for giggles.. i'm rebooting the server. i'm going to see if nmap will portscan on udp |
| 00:38.17 | carrar | nmap will portscan whatever you tell it |
| 00:38.23 | carrar | so yes |
| 00:41.02 | p3nguin | I can't think of any reason you could get successful registration to one provider but not another. |
| 00:42.54 | p3nguin | sip.flowroute.com:5060 1234567 105 Registered Sat, 10 Sep 2011 19:39:53 |
| 00:43.00 | p3nguin | They're working tonight. |
| 00:43.03 | rotten777 | p3nguing are you on linux? |
| 00:43.07 | p3nguin | yes |
| 00:43.38 | rotten777 | i can't tell if my hairpin nat is setup right but i'm not getting the port showing as open |
| 00:46.56 | rotten777 | the port shows open from here... not sure from the outside |
| 00:47.49 | carrar | 216.115.69.144: sip-lv1.flowroute.com |
| 00:47.49 | carrar | PORT STATE SERVICE |
| 00:47.49 | carrar | 5060/udp open|filtered sip |
| 01:01.01 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 01:01.07 | rotten777 | well the softphone works fine |
| 01:01.11 | rotten777 | the asterisk registration doesn't |
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| 01:41.09 | p3nguin | I don't understand that. I can register to flowroute, so it's likely either your configuration or your networking giving issues. |
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| 02:04.18 | carrar | How can that be possible!! |
| 02:04.35 | p3nguin | It's not possible. I made it all up. |
| 02:07.18 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 02:07.20 | Dovid | hi all |
| 02:07.21 | Dovid | res_timing_pthread.so |
| 02:07.37 | Dovid | is that needed of i have res_timing_dahdi.so |
| 02:07.52 | WIMPy | no |
| 02:08.28 | WIMPy | at least not if you can use res_timing_dahdi. |
| 02:08.35 | Dovid | which one is better to use or is it 6 of one, half a dozen of the other? |
| 02:09.01 | Dovid | when i built asterisk it seems to have built both |
| 02:09.26 | WIMPy | If you have dahdi hardware, that's definitely the choice. |
| 02:10.06 | Dovid | i have no dahdi hardware. in that case what's best to use? |
| 02:11.17 | WIMPy | There have been issues with timerfd that can be cured with dahdi dummy timing. |
| 02:11.36 | WIMPy | I'm uding timerfd without issues since it is available. |
| 02:12.19 | Dovid | any where i can see what the difference's pro's/con's of each ? |
| 02:12.52 | WIMPy | If you want meetme or page, you need dahdi anyway. |
| 02:14.19 | Dovid | ok. using dahdi |
| 02:14.23 | Dovid | thanks WIPMy |
| 02:15.55 | WIMPy | There was something about migrating page to confbridge, but I'm not sure if that has been done or is being done. |
| 02:32.48 | p3nguin | Is contrib/scripts/get_mp3_source.sh supposed to be run before make? |
| 02:33.29 | p3nguin | I guess it would make sense that the source needs to be there before make can build the modules, huh? |
| 02:36.01 | *** join/#asterisk DarkStar851 (~DarkStar8@142.163.199.37) |
| 02:36.15 | DarkStar851 | Heh, this channel seems much more lively than #trixbox. |
| 02:36.41 | DarkStar851 | Wouldn't suppose any of you mates know how to forward a PBX extension to a remote SIP address? |
| 02:37.01 | DarkStar851 | Like extension ### forwards to sip:+19999999999@sip.voice.google.com |
| 02:37.39 | p3nguin | There's no forwarding involved... it's a simple Dial(). |
| 02:38.03 | DarkStar851 | So just configure a script for that extension? |
| 02:38.08 | WIMPy | p3nguin: That's what make tells you. |
| 02:38.11 | DarkStar851 | We're using existing SIP phones. |
| 02:38.18 | p3nguin | exten => ###,1,Dial(SIP/19999999999@sip.voice.google.com) |
| 02:38.42 | DarkStar851 | Neat, I figured there'd be something more involved than that. Thanks p3nguin. I'm trying to figure out our systems for work. |
| 02:39.25 | p3nguin | wimpy: If make says to run the script, do you think it means to run the script and then run make again? 'Cause I'm pretty sure it does not say to run it again. |
| 02:39.49 | WIMPy | yes |
| 02:39.51 | DarkStar851 | We've been trying to delegate things off to other services (pbxes.org) so our servers don't flood (small business) and GV seems pretty good for it. |
| 02:39.51 | p3nguin | But if make errors out because the script hasn't been run, it makes sense to run make again after downloading the mp3 source. |
| 02:40.23 | WIMPy | Doesn't it say you should have done so before make? |
| 02:40.38 | p3nguin | I'll try to catch the message when I get there. |
| 02:42.14 | p3nguin | If the script was run before running make, do you think the message will still appear? |
| 02:43.03 | p3nguin | I'm creating a package, so I put in a line to run that script before running make. |
| 02:43.29 | WIMPy | no |
| 03:08.09 | p3nguin | If I'm okay with using mpg123, is there any reason to build rawplayer? |
| 03:08.34 | p3nguin | It's causing a problem, so if I don't need it, that would be great. |
| 03:08.57 | WIMPy | Never heard of rawplayer. |
| 03:09.20 | p3nguin | asterisk-1.8.6.0/contrib/utils/README.rawplayer |
| 03:11.12 | WIMPy | Wouldn't that be the same as using a mono 8Ks/s wav? |
| 03:11.46 | p3nguin | It's an alternative to using mpg123 for playing mp3s in moh. |
| 03:12.12 | WIMPy | And with all those wideband codecs around, do we really want to store the samples as 8ks/s? |
| 03:12.47 | WIMPy | I read about converting them with sox and play the converted file. |
| 03:13.35 | p3nguin | I have no concern about using mpg123 to play mp3, so I guess I'll skip rawplayer and eliminate the need to fix this problem. |
| 03:14.03 | p3nguin | I didn't have this problem when I built a 1.8.5.0 package, so I don't know what the difference is now. |
| 03:14.46 | p3nguin | I don't remember if I installed format_mp3 or not in the 1.8.5.0 package, but I did in this 1.8.6.0 package. Maybe that has something to do with it. |
| 03:14.52 | p3nguin | or maybe it doesn't. |
| 03:19.30 | p3nguin | Oh, the 1.8.5.0 package does not include rawplayer. Not sure how it failed but continued when this build failed and exited. Must be change in my PKGBUILD. |
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| 06:54.10 | ChannelZ | quietly picks his nose |
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| 07:48.07 | KNERD | I am not getting sound if connecting to the server from the public IP address |
| 07:49.50 | mtbf | Looks like a problem with RTP, check your firewall and port forwarding settings for used RTP ports range. |
| 07:57.37 | KNERD | did that already |
| 08:01.23 | KNERD | 5060 10000-20000 |
| 08:02.50 | KNERD | 5222 |
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| 08:21.50 | mtbf | KNERD: in my firewall i use range 5000-60000 (output porst, for client), previously it was also so tight and I continously experienced no voice from time to time, cause the VoIP providers were changing it so try it, unless you're sure this is the range used by your server, check the firewall logs. |
| 08:22.14 | ChannelZ | which side isn't getting sound? Neither? |
| 08:22.32 | KNERD | correct, neither |
| 08:22.49 | ChannelZ | Is one behind NAT? |
| 08:23.00 | KNERD | mtbf: if I use below 10000 it starts affecting other services, such as video streaming |
| 08:23.07 | KNERD | ChannelZ: yes, NAT |
| 08:23.14 | ChannelZ | Does Asterisk know that? |
| 08:23.19 | mtbf | ;D |
| 08:23.49 | KNERD | if I remember correctly I had set it up in the settings...been a while |
| 08:24.06 | ChannelZ | Well who is behind NAT? Asterisk or the device trying to connect to it? |
| 08:24.09 | KNERD | extern -...bla bal |
| 08:24.16 | KNERD | Asterisk |
| 08:24.53 | ChannelZ | ok.. so yes in sip.conf you need "externip" set correctly, and "localnet" so it's able to tell the inside network apart from the outside |
| 08:25.33 | KNERD | let me take a peek |
| 08:25.54 | ChannelZ | Then you need ports 5060 and some range (as specified in rtp.conf) open and port-forwarded to the Asterisk box if the NAT router isn't snooping in on SIP to do it its self |
| 08:28.39 | KNERD | I have dynamic IP. though it does not change unless I lose power to interner box |
| 08:29.04 | ChannelZ | well externip needs to be correct for whatever your IP is at the moment |
| 08:29.25 | ChannelZ | Because your Asterisk box has a fake IP address, it has to know how to lie to the other guy and tell them the correct IP to send their RTP to |
| 08:32.26 | KNERD | strange..my system has issues.."read on file system" for whatever reason |
| 08:32.49 | ChannelZ | you mean 'read only'? |
| 08:32.50 | KNERD | *only |
| 08:32.59 | ChannelZ | That's not good |
| 08:33.21 | KNERD | i guess time to wipe drive |
| 08:33.49 | ChannelZ | Did it fail an fschk or something? |
| 08:34.01 | KNERD | yes, so I had to run it manually |
| 08:34.09 | KNERD | been having brownouts lately |
| 08:34.30 | ChannelZ | hmm. What filesystem is it running? |
| 08:34.44 | KNERD | dont kmow |
| 08:35.24 | KNERD | ext2f |
| 08:36.31 | ChannelZ | hmm. Well if you do rebuild I'd use at least ext3 if your system is prone to having the plug pulled a lot like that |
| 08:36.37 | ChannelZ | or some other journaled filesystem |
| 08:36.46 | ChannelZ | brb potty break |
| 08:36.51 | KNERD | ok |
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| 09:58.08 | Dovid | hi all |
| 09:59.28 | Dovid | I have two machines running 1.8.x when I do: asterisk -rx "core show sysinfo" |
| 09:59.46 | Dovid | one machine i get the same as 1.6. on the other I get more info. i get two extra lines. any idea hwy? |
| 09:59.48 | Dovid | why* |
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| 10:30.17 | kaldemar | Dovid: by all means, tell what the lines are. |
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| 14:37.24 | rotten777 | anyone have any idea why i can make a call but no audio is coming through? |
| 14:37.35 | rotten777 | is that the rtp forwarding to asterisk behind nat? |
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| 14:43.18 | loconut | hello- I have a question about the queue log format. I read voip-info's breakdown of the queue log, but it doesn't say whether calltime includes holdtime eg talk time = calltime - holdtime. I'd been assuming this, but today I had a report i generated with negative time since calltime was less than holdtime. So now I'm wondering. |
| 14:44.07 | loconut | this would be for COMPLETE* events |
| 14:49.45 | loconut | can any one tell me where this is properly documented without reading source code? |
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| 15:03.11 | loconut | found in source |
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| 15:09.43 | Korolev | Hi guys, I've been googling aroung but can't find an answer to this |
| 15:10.22 | Korolev | is there a way to make asterisk respond with circuit busy if it reaches a certain amount of calls per second? |
| 15:13.15 | Korolev | I have a server for USA termination, under very heavy load and asterisk seems to break around 30 cps. The idea is to have it return 503 so I can pass the call to another server |
| 15:13.51 | WIMPy | You can set a load maximum. |
| 15:14.05 | Korolev | yeah, but max load is not really helping |
| 15:14.24 | Korolev | the host is a dual quad core, 8 gb of ram |
| 15:14.44 | Korolev | so even though it is very heavy load for asterisk to handle |
| 15:14.54 | Korolev | the cpu is not really sweating at all |
| 15:15.44 | Korolev | and the intention is to chroot asterisk and bind it to different ip addresses in the same server |
| 15:16.13 | Korolev | so I can run 4 asterisk in the same server, to handle all the calls. the limitation seems to be asterisk itself, not the hardware |
| 15:19.33 | cyford | at that rate you should look into openser and loadbalance the asterisk servers |
| 15:23.33 | Korolev | I considered that, but I was hoping there was something I was missing in asterisk, so I could sove it without introducing another piece of software that I dont know anything about :) |
| 15:23.57 | Korolev | looks like im out of options |
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| 15:55.46 | cyford | are you using realtime? |
| 15:57.26 | cyford | stupid question, does asterisk load the dialplan in memory, or does it get it from the config every call? |
| 15:59.02 | WIMPy | What do you think 'dialplan reload' is there for? ;-) |
| 15:59.20 | cyford | well i said it was a stupid question...lol |
| 15:59.39 | cyford | i thought about it once after i asked |
| 15:59.42 | cyford | lol |
| 16:00.22 | cyford | so having a faster hard drive would only improve reloding asterisk right? |
| 16:00.40 | WIMPy | How big is your dialplan? |
| 16:01.03 | cyford | pretty big |
| 16:01.30 | cyford | but its not for me |
| 16:01.31 | WIMPy | How many MB/s will a slow HD read? |
| 16:02.34 | cyford | my asterisk is installed over a san.... with 4 gbs links... |
| 16:02.42 | cyford | im not having any issues |
| 16:03.03 | cyford | fishing for korolev |
| 16:03.31 | WIMPy | dynamic realtime would certainly make a difference here. |
| 16:03.31 | Korolev | my dialplan? |
| 16:03.54 | Korolev | not really that big, the only costly function is curl to retreive routing info and to hangup |
| 16:04.01 | WIMPy | But for a normal dialplan, a floppy disk should be good enough. |
| 16:04.05 | Korolev | let me post it so you guys can see it |
| 16:04.38 | cyford | im using freepbx |
| 16:04.40 | cyford | lol |
| 16:10.01 | Korolev | http://www.sourcepod.com/ciltxl72-5484 |
| 16:10.05 | Korolev | there, thats the whole thing |
| 16:10.31 | Korolev | the ... between DIAL-CHANUNAVAIL and h is for every dialstatus |
| 16:10.44 | Korolev | they are pretty much the same thing as whats in CHANUNAVAIL |
| 16:11.03 | Korolev | except for busy, noanswer, dontcall and cancel |
| 16:11.51 | WIMPy | Have you tried to time the CURL call? |
| 16:12.19 | Korolev | yes, I stress tested the webservices to 172 calls per second |
| 16:12.27 | Korolev | way above what I get from asterisk |
| 16:13.38 | WIMPy | How? Maybe it's the call that takes time? |
| 16:14.26 | Korolev | the call setup? |
| 16:14.43 | Korolev | sure, it takes second, sometimes up to a minute in ringing state |
| 16:15.06 | WIMPy | No the CURL call. |
| 16:15.30 | Korolev | the servers are next to each other, on a 100 mbps link |
| 16:15.38 | Korolev | ping is < 1ms |
| 16:16.21 | Korolev | asterisk and the apache serving the webservices |
| 16:16.24 | WIMPy | I guess there's a fork involved. That may take some time. |
| 16:17.31 | Korolev | when I get too many calls per second, what I see is a lot of calls ringing, but the legs dont exist, either from the customer or the carrier |
| 16:18.04 | Korolev | they begin to fill up until it hits 500 calls, which is the max I set asterisk to |
| 16:18.39 | Korolev | and a bit after that, asterisk stops responding to cli commands |
| 16:19.07 | Korolev | cpu load is negligible |
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| 16:20.06 | WIMPy | That somehow smells liek some sort of I/O issue. Do you have debug logging enabled or something? |
| 16:21.04 | olinux | anyone have a recommended hosting provider to host my switchvox/asterisk box? |
| 16:22.24 | Korolev | no, i disabled loging and cdr |
| 16:22.35 | cyford | some of my clients are using vitelity.net |
| 16:22.49 | cyford | 50 megs up an down |
| 16:22.59 | cyford | 125 for 6 servers |
| 16:23.12 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 16:23.45 | Korolev | asterisk is running in highpriority |
| 16:24.16 | Korolev | max files is huge, but it doesnt open more than 5 or 6 files per active call |
| 16:24.32 | olinux | 125 megs? |
| 16:25.11 | Korolev | thats why I started using curl, I was doing agi and thought the problem might be in starting and tearing down a new php process per call |
| 16:25.48 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 16:26.40 | Korolev | curl does seem to help a bit, but not much |
| 16:30.00 | Korolev | strangely |
| 16:30.06 | Korolev | a P4, 1 Gb of ram |
| 16:30.11 | Korolev | seems to perform better |
| 16:30.18 | Korolev | with the same OS and the same version of asterisk |
| 16:30.46 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 16:31.00 | p3nguin | One gigabit of ram... that's a bunch! |
| 16:31.15 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 16:32.32 | WIMPy | Byte is probably wrong anyway. Or are modern CPUs still able to address a set of 8 bits? I don't think so. |
| 16:35.03 | *** join/#asterisk bmg505 (~leon@196-209-120-60.dynamic.isadsl.co.za) |
| 16:36.15 | florz | if you consider current x86 a "modern CPU" ... well, it's called x86 for a reason ;-) |
| 16:37.22 | WIMPy | I'm pretty sure it's only emulation. |
| 16:38.26 | florz | well, that obviously depends on what you mean by "emulation" |
| 16:38.39 | Korolev | these are very large bits |
| 16:39.12 | florz | at the ISA level, you can address bytes, and as the ISA is implemented by the CPU, the CPU can address bytes |
| 16:39.20 | WIMPy | If I store an 8-bit value, that's probably translated to a read-modify-write of at least 32 bits, or probably even more. |
| 16:40.17 | florz | well, that depends on the level you are looking at |
| 16:40.18 | WIMPy | That's I/O. I'm just on to RAM. |
| 16:40.35 | florz | no, ISA is also the Instruction Set Architecture |
| 16:41.06 | Korolev | christ, my gigabits contain 256 milibits each |
| 16:41.09 | *** join/#asterisk gravin (~gravin@125.70.50.60.brf01-home.tm.net.my) |
| 16:41.17 | Korolev | can we get back to my calls per second? :D |
| 16:41.22 | florz | the ISA bus isn't implemented by the CPU anymore these days (and actually not at all on current mainboards any more) |
| 16:41.53 | WIMPy | Korolev: IT usual uint it call attpempts per second :-) |
| 16:42.18 | WIMPy | s/IT/The/ |
| 16:43.07 | florz | and at the RAM interface, accesses usually happen in cache lines, so that would be 32 or 64 bytes(!) naturally aligned |
| 16:43.09 | Korolev | uint seems like a waste of space for such a small number |
| 16:43.14 | Korolev | :) |
| 16:43.54 | WIMPy | florz: That's what I thought, but the cache could be disabled. |
| 16:44.13 | p3nguin | Hmm. I wonder... |
| 16:44.18 | WIMPy | Korolev: Äh, "unit" |
| 16:44.21 | p3nguin | I did it right. |
| 16:44.25 | p3nguin | s/did/really/;s/it/messed/;s/right/up/ |
| 16:44.42 | p3nguin | Nope, doesn't work. |
| 16:45.37 | Korolev | comma maybe? |
| 16:48.33 | florz | WIMPy: in that case I think the RAM accesses should still be byte accesses - after all, all higher layers need to be able to do uncached byte accesses anyhow, as most I/O nowadays happens through memory mapped I/O, so as far as the computation core is concerned, there is no difference between a write to a PCI card's mmapped I/O register and a RAM write, and in the former case, read-modify-write cannot be used, as writes (and even reads) can have side effe |
| 16:49.50 | WIMPy | Yes, but I wouldn't have exprected that to still work for RAM. |
| 16:51.15 | florz | well, if it doesn't, then the memory controller would have to implement some read-modify-write buffer (you can operate on RAM with cache disabled, after all) |
| 16:52.24 | florz | and even then it's questionable if you really want to consider the memory controller part of the CPU in that context, even though it nowadays tends to be on the CPU die ;-) |
| 16:52.52 | WIMPy | That's definitely debatable. |
| 16:53.18 | WIMPy | Carappy compatibility stuff :-( |
| 16:53.28 | florz | heh :-) |
| 17:01.48 | florz | plus, given that SDRAM chips have to have a buffer at their sense amplifier anyhow, and that DIMMs tend not to have (DRAM) rows the same size as the platform's cache line, the only real advantage of not implementing byte writes there would be saving a few lines at the DIMM interface |
| 17:03.18 | florz | I just looked it up: DDR3 still has data mask lines for selecting bytes to be read/written |
| 17:03.48 | florz | so I guess read/modify write in uncached operation indeed does happen inside the DRAM chip |
| 17:04.12 | WIMPy | Ok, so on x86 the bytes haven't grown, yet. |
| 17:04.39 | florz | so it seems :-) |
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| 18:44.42 | simplydrew | p3nguin: are you around? |
| 18:44.51 | p3nguin | yes |
| 18:45.38 | simplydrew | p3nguin: was wondering if you could assist me a little further with chan_sccp. I'm still hitting a brick wall with this error |
| 18:45.58 | p3nguin | There's nothing I can really do about a build error. |
| 18:46.04 | simplydrew | the asterisk header files are there in the correct directory, so I don't believe that's the source of the problem |
| 18:48.53 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:51.12 | p3nguin | I use asterisk 1.4, and I don't have any problems building chan_sccp 2, 3.0, or 3.1 on it. |
| 18:52.06 | p3nguin | I use 3.0 on it. |
| 18:53.06 | p3nguin | I built an asterisk 1.8.6.0 package last night that I may or may not install later today. If chan_sccp 4.0 will build against it, I may start using that pair. |
| 18:53.46 | p3nguin | The big thing is that it has to be a completely seamless change-over. If there is anything different, that'll be a problem. |
| 18:55.13 | WIMPy | Is that made for 1.8 then? |
| 18:55.41 | p3nguin | The 4.0 dev branch is supposed to now include support for asterisk 1.8. |
| 18:56.48 | WIMPy | And for 10 it needs some rewrites again. |
| 18:57.10 | p3nguin | I'm not sure where they plan to go next. |
| 18:57.51 | WIMPy | That goes for a handfull of channels. :-( |
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| 19:29.19 | *** join/#asterisk zamba (marius@flage.org) |
| 19:29.39 | zamba | i want to define one user as an admin for a meetme conference room.. how do i do this? |
| 19:29.52 | zamba | does the user have to call a specific number for that to work? |
| 19:30.14 | zamba | or can it be a pin code or something to ident yourself as an admin? |
| 19:31.54 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 19:42.30 | blitzrage | zamba: it can be anything you want -- you'll need to define something like a separate extension, and then add a pin (if you want), and once authenticated, call MeetMe() with the appropriate flag for administrator |
| 19:43.16 | blitzrage | the 'a' option to MeetMe() is 'admin' |
| 19:43.16 | *** join/#asterisk cyford (~cyford@c-24-99-17-185.hsd1.ga.comcast.net) |
| 19:43.25 | zamba | yeah, but then everyone became admins :) |
| 19:43.35 | blitzrage | only if they call that extension and authentication with the pin |
| 19:43.38 | blitzrage | have a separate extension for users |
| 19:43.59 | zamba | ah, ok |
| 19:44.05 | blitzrage | alternatively, allow someone while in the conference to dial a DTMF digit, exit to another menu which requires the correct pin, then rejoin the conference as the admin |
| 19:44.27 | zamba | hm, i'm not fluent enough in dialplan hacking to accomplish that :) |
| 19:44.41 | blitzrage | zamba: look at the p() option to meetme |
| 19:44.48 | blitzrage | it'd be pretty easy |
| 19:45.57 | zamba | nah, giving that up.. i understand nothing of this and it makes my brain hurt :) |
| 19:46.06 | blitzrage | ouch |
| 19:46.14 | blitzrage | anyways, make 2 separate extensions then |
| 19:46.23 | zamba | http://lists.digium.com/pipermail/asterisk-users/2008-February/206063.html |
| 19:46.24 | zamba | i see that example |
| 19:46.32 | blitzrage | put authentication on one, and have that authenticated user join as the admin |
| 19:46.34 | zamba | but that has [conf] |
| 19:46.44 | WIMPy | Stay away from doctors then. They will remove anything that hurts. |
| 19:46.47 | zamba | a separate context |
| 19:46.58 | blitzrage | there are many ways to accomplish the same thing :) |
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| 20:14.36 | *** join/#asterisk BladeMcCool (~BladeMcCo@d209-121-230-44.bchsia.telus.net) |
| 20:16.09 | BladeMcCool | i have ubuntu 10.04LTS, installed askterisk via apt-get (works) but cannot find package asterisk-respeak .. what am i doing wrong? apt-get install asterisk-respeak -> Couldn't find package asterisk-respeak ... i dont know what repo i need to add or how to add it? (or how to see what repos I have ??) |
| 20:16.24 | BladeMcCool | *asterisk-espeak i meant to put. |
| 20:16.53 | BladeMcCool | this site shows it for ubuntu: https://launchpad.net/ubuntu/+source/asterisk-espeak but i dunno if that is relevant |
| 20:30.35 | BladeMcCool | i tried apt-get source asterisk-espeak and same error basically, 'Unable to find a source package for asterisk-espeak' ... what am i doing wrong? how do i make it find the right repo or whatever? |
| 20:49.43 | blitzrage | no idea what asterisk-espeak is |
| 20:49.57 | blitzrage | sounds like a third party package, so likely you're missing the apt source |
| 20:54.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:54.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 21:07.48 | BladeMcCool | right. ... well any text to speech i guess i need. something easy to install on lucid. lol. dont really care what it sounds like just need quick voice menus |
| 21:15.14 | BladeMcCool | lool sorry for cluttering your channel. step 1) recognizing that there is no prebuilt for lucid, step 2) finding source and actually installing all dependencies (yay). 3) realizing that source is out of date and finding newsest one .. 4) win :D |
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| 21:38.48 | simplydrew | p3nguin: ah gotcha. I was curious as to if you had tried the 1.8.6.0 package yet or not |
| 21:38.57 | simplydrew | just not sure where I'm going wrong with this. very strange. |
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| 22:28.12 | ChannelZ | BladeMcCool: A mic/phone and Record() are your friend |
| 22:29.29 | ChannelZ | Or go to cepstral.com and type in demo text and save off the wav files it makes :P |
| 22:31.10 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 22:31.18 | sequencer | hi all :) |
| 22:31.27 | ChannelZ | aloha |
| 22:31.30 | sequencer | hey |
| 22:32.03 | sequencer | am trying to run asterisk behind NAT but am having some trouble with that, am trying to register to an external SIP provider with no hope :s |
| 22:33.50 | ChannelZ | you need to set externip and localnet in sip.conf |
| 22:34.17 | ChannelZ | And probably need to port-forward 5060 and whatever port range you use from rtp.conf to your asterisk box |
| 22:34.18 | sequencer | hmm.. lets try , thanks |
| 22:38.10 | rotten777 | anyone know why my caller-id isn't working when i call out? |
| 22:39.13 | sequencer | am still having 0 registration :s |
| 22:39.35 | rotten777 | sequencer are you in the terminal for asterisk? |
| 22:39.41 | sequencer | yes |
| 22:39.46 | rotten777 | sip show registry |
| 22:39.51 | rotten777 | does it say reg sent? |
| 22:40.09 | sequencer | 0 SIP registrations. |
| 22:40.14 | sequencer | thats what i have :s |
| 22:40.14 | rotten777 | sip show peers |
| 22:40.33 | sequencer | 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] |
| 22:40.47 | rotten777 | the 1 peer is what |
| 22:41.04 | sequencer | its the sip provider am tring to peer with |
| 22:41.24 | sequencer | basically my SIP trunk that i use to make calls through |
| 22:41.39 | rotten777 | whats your register statement in the sip.conf |
| 22:41.48 | rotten777 | sans the password obviously |
| 22:41.56 | sequencer | the register is correct |
| 22:42.07 | sequencer | because i took it from the other server am using |
| 22:42.13 | sequencer | but this one is behind NAT |
| 22:42.19 | rotten777 | you have nat=yes below it |
| 22:42.21 | rotten777 | in general |
| 22:42.28 | rotten777 | ? |
| 22:43.26 | rotten777 | do you have externhost=something.wherever.com ? or an outside ip? |
| 22:43.34 | sequencer | i had it with the peer definition |
| 22:43.40 | sequencer | let em check |
| 22:44.08 | sequencer | I have these externip=196.218.255.29 localnet=192.168.1.3 |
| 22:44.09 | rotten777 | general section defines the asterisk server... if it is behind nat then you put nat=yes there |
| 22:44.21 | rotten777 | localnet=192.168.1.0/255.255.255.0 |
| 22:44.25 | sequencer | oh |
| 22:44.30 | rotten777 | the peer isn't behind nat i'm assuming |
| 22:44.33 | rotten777 | right? |
| 22:44.49 | sequencer | the per am trying to connect with isnt behind nat |
| 22:45.02 | rotten777 | ok then in the context for that peer, do not have nat=yes |
| 22:45.12 | sequencer | oh ok |
| 22:45.32 | rotten777 | i'm an expert. i've been successfully using my asterisk server for about 10 hours now lol |
| 22:45.43 | rotten777 | i went through the same thing you're doing though |
| 22:45.51 | sequencer | i have nat=auto in the context |
| 22:45.57 | rotten777 | for the peer? |
| 22:45.59 | sequencer | yes |
| 22:46.01 | rotten777 | nat=no |
| 22:46.19 | rotten777 | nat=yes ;goes under general |
| 22:46.26 | sequencer | alright |
| 22:46.33 | sequencer | heres what i did |
| 22:46.33 | rotten777 | once those changes are made sip reload |
| 22:46.43 | sequencer | asterisk is set to the DMZ |
| 22:46.46 | rotten777 | ok |
| 22:46.59 | sequencer | i made port forwarding for 10000 - 20000 to * box |
| 22:47.24 | rotten777 | i would turn off the dmz, just port forward UDP 5060 and UDP 10000 - 20000 |
| 22:48.00 | *** join/#asterisk DarkStar851 (~DarkStar8@142.163.169.117) |
| 22:48.02 | rotten777 | make sure it is UDP and not TCP |
| 22:48.15 | sequencer | cant i set to Both ? |
| 22:48.27 | DarkStar851 | *bangs head loudly against wall* Can anyone complete a sip call to darkstar851-200@pbxes.org? |
| 22:48.30 | sequencer | as i have that opyon |
| 22:48.35 | rotten777 | sure |
| 22:48.43 | rotten777 | darkstar851: i can try |
| 22:48.55 | DarkStar851 | That would be awesome rotten777, I think it's my settings. |
| 22:49.01 | DarkStar851 | That or PBXes is blocking something. |
| 22:49.09 | sequencer | <--- SIP read from UDP:74.217.82.210:5060 ---> SIP/2.0 404 Not Found Am getting this :s < |
| 22:50.26 | rotten777 | 404 not found for sip? :X |
| 22:50.44 | sequencer | :s is it that bad ? :s |
| 22:51.01 | rotten777 | darkstar i can't dial that for some reason |
| 22:51.07 | rotten777 | my softphone isn't happy |
| 22:51.12 | DarkStar851 | Nor is mine. |
| 22:51.16 | DarkStar851 | Stupid PBXes. :\ |
| 22:51.20 | rotten777 | sequencer that is bad |
| 22:51.30 | rotten777 | sequencer: what kind of router is it |
| 22:51.38 | sequencer | oh god |
| 22:51.53 | sequencer | its a sagem router+DSL Modem |
| 22:52.00 | rotten777 | eek |
| 22:52.03 | sequencer | comes from my phone telco |
| 22:52.08 | rotten777 | yeah.... |
| 22:52.15 | rotten777 | do you run windows or linux? |
| 22:52.32 | sequencer | my box is winxp , * is on CENTOS |
| 22:53.15 | sequencer | the router is configured to forward many ports to different servers in the office |
| 22:53.34 | sequencer | so i cant replace the router, i know that my settings were correct when i had a box without NAT |
| 22:53.53 | ChannelZ | looks like your ITSP is rejecting you |
| 22:53.59 | rotten777 | ok so basically you're getting a 404 when you're dialing out? |
| 22:54.11 | sequencer | this is not when am dialing |
| 22:54.23 | sequencer | this is whenam registering with 102 NOTIFY |
| 22:54.59 | rotten777 | yeah if you're registering then the itsp is saying that user isn't there |
| 22:55.10 | rotten777 | so it sounds like your register statement |
| 22:55.12 | rotten777 | which itsp? |
| 22:55.25 | sequencer | aptela |
| 22:55.50 | sequencer | one sec.. i found sth weird |
| 22:55.52 | rotten777 | do they give you a peer sample in your profile? |
| 22:56.02 | sequencer | From: "asterisk" <sip:asterisk@voip.aptela.com> |
| 22:56.21 | sequencer | this should be my actual username :s |
| 22:56.34 | rotten777 | username:password@host |
| 22:56.35 | rotten777 | yes |
| 22:56.57 | sequencer | username = is deprecated ? |
| 22:57.02 | sequencer | in the context |
| 22:57.08 | rotten777 | defaultuser |
| 22:57.12 | sequencer | alright |
| 22:57.15 | rotten777 | defaultuser=sequencer |
| 22:57.54 | sequencer | looks better |
| 22:57.59 | sequencer | rom: "asterisk" <sip:ext110.mecmc@voip.aptela.com>;tag=as26ad30e3 |
| 22:58.38 | sequencer | still doesnt regsiter though :s |
| 22:58.39 | rotten777 | sip show registry |
| 22:58.46 | sequencer | 0 SIP registrations. |
| 22:58.55 | rotten777 | sip set debug on |
| 22:59.09 | rotten777 | does it show registration sent? |
| 22:59.09 | sequencer | nothing shows up |
| 22:59.15 | sequencer | no activity |
| 22:59.25 | rotten777 | can you pastebin your sip.conf (removing passwd) |
| 22:59.32 | sequencer | sure |
| 22:59.39 | DarkStar851 | :\ After adding an inbound for my SIP address it's giving me a GTFO signal when I call. |
| 23:00.14 | sequencer | register => ext110.mecmc:CCCCCCCCC@voip.aptela.com |
| 23:00.18 | rotten777 | DarkStar851, you're calling out gives you a go away? |
| 23:00.33 | DarkStar851 | I'm calling to a SIP address, |
| 23:00.38 | DarkStar851 | that one at PBXes. |
| 23:00.50 | DarkStar851 | I added an inbound route for it on PBXes service and now it's giving me a go away. |
| 23:01.00 | rotten777 | ~pastebin |
| 23:01.00 | infobot | [~pastebin] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:01.17 | sequencer | rotten777 i know pastebin |
| 23:01.19 | rotten777 | sequencer use pastebin.com and post your whole sip.conf without passwords there's something else going on |
| 23:01.20 | rotten777 | lol |
| 23:01.21 | sequencer | its just one line |
| 23:01.29 | sequencer | oh |
| 23:01.52 | sequencer | that would be sip.conf and users.conf |
| 23:01.52 | rotten777 | no just sip.conf now |
| 23:02.00 | sequencer | alright one sec |
| 23:02.00 | rotten777 | dialplan and etc. shouldn't matter for just initial sip registration |
| 23:02.11 | sequencer | dialplan is extensions.com |
| 23:02.14 | sequencer | dialplan is extensions.conf* |
| 23:02.20 | rotten777 | yup |
| 23:02.28 | sequencer | but users.conf has my sip cotnext |
| 23:02.41 | rotten777 | what is in sip.conf |
| 23:02.55 | sequencer | just the [general] and the regsiter string |
| 23:03.10 | sequencer | register => ext110.mecmc:CCCCCCCCC@voip.aptela.com |
| 23:03.51 | rotten777 | i've never seen that before ... it doesn't have the [general] with a ton of other lines? |
| 23:04.07 | sequencer | mostly are commented out |
| 23:04.11 | sequencer | one sec ill pate em |
| 23:04.30 | rotten777 | k |
| 23:05.54 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 23:06.04 | sequencer | this is my sip.conf |
| 23:06.05 | sequencer | http://pastebin.com/SSx0PXRt |
| 23:06.11 | sequencer | without the commented text |
| 23:06.58 | rotten777 | your register statement needs to be on line 10 after localnet |
| 23:07.03 | rotten777 | you've got it outside of the general context |
| 23:07.21 | sequencer | oh does it matter ? :s |
| 23:07.31 | rotten777 | yes it does ;) |
| 23:08.03 | sequencer | it does! |
| 23:08.13 | sequencer | 105 Registered! |
| 23:08.17 | rotten777 | :) |
| 23:08.26 | sequencer | lets try to make a call |
| 23:08.39 | sequencer | oh wait.. |
| 23:08.44 | sequencer | no users :s |
| 23:08.46 | sequencer | one sec ;) |
| 23:12.17 | sequencer | i changed my username in sip.conf tho the actual one i want to use |
| 23:12.40 | sequencer | but it just says 120 Request Sent |
| 23:13.03 | sequencer | <PROTECTED> |
| 23:13.55 | rotten777 | you're sure voip.aptela.com is the sip server? |
| 23:14.07 | sequencer | yes |
| 23:14.16 | sequencer | on the older username it worked |
| 23:14.21 | sequencer | i just changed it to the new one |
| 23:16.57 | rotten777 | sip show peers |
| 23:18.58 | sequencer | i used the old username and it worked fine |
| 23:19.03 | sequencer | now am trying to make a call |
| 23:19.27 | sequencer | Call from '001' (192.168.1.2:51648) to extension '7043156026' rejected because extension not found in context 'defaultOut'. |
| 23:19.48 | sequencer | defaultOut is defined in extensions.conf |
| 23:20.21 | sequencer | it worked now.. |
| 23:20.31 | sequencer | silly, forgot to have _ befroe XXX's :s |
| 23:22.10 | rotten777 | ahh ok cool so you're all working? |
| 23:22.20 | rotten777 | incomgin outgoing? |
| 23:22.29 | sequencer | outgoing |
| 23:22.38 | sequencer | am not sure if i want to setup incoming yet |
| 23:23.20 | sequencer | i need to see if i can register from an outside peer |
| 23:23.45 | rotten777 | registering is just port 5060 udp being forwarded |
| 23:24.12 | rotten777 | getting audio and making sip sessions activate are 2 separate things |
| 23:24.19 | rotten777 | you have the 10000-20000 setup? |
| 23:24.23 | rotten777 | the forward? |
| 23:29.32 | sequencer | yes |
| 23:29.49 | sequencer | am getting 408 request tmeout on the external phone |
| 23:30.00 | sequencer | even though its registered |
| 23:30.18 | rotten777 | hmm... registration is hitting your asterisk but the ack isn't getting back? |
| 23:30.58 | sequencer | 001/001 (Unspecified) D N 0 Unmonitored |
| 23:31.29 | rotten777 | yeah it isn't registering |
| 23:32.02 | rotten777 | going for a cigar.. i will return if you still need help |
| 23:32.11 | sequencer | thanks man |
| 23:32.54 | sequencer | says unauthorized : |
| 23:34.45 | sequencer | it registered |
| 23:34.54 | sequencer | but when making a call its unauthorized :s |
| 23:37.52 | sequencer | http://pastebin.com/mCrrF76p |
| 23:41.51 | sequencer | rotten777 you there ? |
| 23:45.13 | sequencer | i set the box as a DMZ and it worked :s |
| 23:51.24 | *** join/#asterisk mindCrime_ (~chatzilla@static-50-52-147-222.drhm.nc.frontiernet.net) |
| 23:53.58 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |