| 00:02.50 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 00:06.41 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 00:16.34 | corretico | <[TK]D-Fender>hey. The problem was on the Cisco configuration. The traffic between the cisco and asterisk are OK |
| 00:16.50 | p3nguin | Still having problems with quoting, I see. |
| 00:16.56 | p3nguin | You're surely not that dense. |
| 00:24.23 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 00:25.45 | JunK-Y | mooO! |
| 00:32.02 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 00:39.45 | [TK]D-Fender | JunK-Y: Salut |
| 00:40.27 | pdtpatrick | Question .. is Asterisk moving towards AEL or LUA ? |
| 00:40.34 | WIMPy | Hi [TK]D-Fender. did you feel homesick? :-) |
| 00:40.47 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 00:41.12 | p3nguin | pdtpatrick: I wouldn't think so. |
| 00:41.20 | pdtpatrick | ? |
| 00:41.22 | [TK]D-Fender | WIMPy: Tough question... felt a lot of things really, not sure I've processed them all yet. |
| 00:41.27 | p3nguin | You may certainly use either or both of those if you wish, though. |
| 00:41.52 | pdtpatrick | right but i don't want to get very familiar with AEL and then they drop it in favor of LUA |
| 00:42.51 | [TK]D-Fender | pdtpatrick: AEL is and always has been a intperted lay over basic dialplan offering less than you could do by yourself and making debugging a greater challenge in trade for a syntax that might feel more familiar to some |
| 00:43.48 | [TK]D-Fender | interpreted* |
| 00:44.33 | [TK]D-Fender | pdtpatrick: As long as it's maintained it will probably have its small following, but they are a considerably smaller statistic |
| 00:45.02 | [TK]D-Fender | PBX_UA is a lot more integrated from little I recall and probably isn't on the same level as AEL |
| 00:46.55 | pdtpatrick | you mean PBX_LUA ? |
| 00:48.10 | [TK]D-Fender | yes |
| 00:48.18 | pdtpatrick | [TK]D-Fender, just so i understood what you said.. AEL is superb compared to the plain dialplan but inferior to lua? |
| 00:49.11 | [TK]D-Fender | pdtpatrick: That is not what I said |
| 00:49.41 | [TK]D-Fender | pdtpatrick: AEL compiles back to common dialplan logic, but in so doing can only offer you less that you could have done in standard dialplan logic yourself |
| 00:50.19 | [TK]D-Fender | pdtpatrick: One immediate key shortcoming was lack of presence support |
| 00:51.11 | pdtpatrick | presence as in JabberStatus ?? |
| 00:51.17 | pdtpatrick | that sucks |
| 00:51.19 | [TK]D-Fender | pdtpatrick: as in basic hints. |
| 00:51.47 | pdtpatrick | okay how about lua from your experience? |
| 00:51.49 | [TK]D-Fender | pdtpatrick: Like the magical "I wan't my phone to light up when my coworker is on a call" |
| 00:51.49 | pdtpatrick | thanks in advance btw |
| 00:51.57 | [TK]D-Fender | want* |
| 00:52.47 | [TK]D-Fender | I am uncertain for LUA. No direct experience with it. It'd be something to look at, but * still revolves around certain key processing concepts so I'm not sure where its "catch" is at. |
| 00:53.36 | [TK]D-Fender | pdtpatrick: before branching out, how has your own experience with the basics of * been going, and what leads you to look at these other two? |
| 00:53.52 | pdtpatrick | well i want to do something like this |
| 00:53.56 | pdtpatrick | https://reviewboard.asterisk.org/r/88/ |
| 00:54.35 | pdtpatrick | currently i'm using the gotoif |
| 00:54.41 | d_preston215 | I know its late as hell, but I just want to say that trixbox royally sucks. |
| 00:54.48 | pdtpatrick | and getting information from jabberstatus |
| 00:55.29 | pdtpatrick | and thought it would make a lot more sense to use something like AEL with case (just like bash scripts) |
| 00:55.49 | pdtpatrick | or use LUA with if (like other languages: python etc) |
| 00:56.04 | [TK]D-Fender | d_preston215: I'm sure others will have told you the same thing repeatedly already. |
| 00:56.32 | [TK]D-Fender | pdtpatrick: do "dialplan show" at * CLI and see what that code of yours turned into. |
| 00:57.07 | [TK]D-Fender | pdtpatrick: And I don't see this "gotoif" you were referring to in there |
| 00:57.50 | [TK]D-Fender | pdtpatrick: I see a single "case" section which is functionally a GotoIF (especially when you look at how it gets parsed) |
| 00:57.51 | pdtpatrick | [TK]D-Fender, here's an exampleexten => s,1,jabberstatus(asterisk,${ARG2},STATUS) |
| 00:57.58 | pdtpatrick | then i do |
| 00:58.00 | pdtpatrick | exten => s,2,gotoif($[$[${STATUS}]<3]?available:unavailable) |
| 00:58.29 | pdtpatrick | but the other link ur using uses switch + case |
| 00:59.05 | pdtpatrick | coming from a python/bash background .. the switch/case construct makes a lot more sense |
| 00:59.11 | pdtpatrick | and looks easier to read |
| 00:59.18 | [TK]D-Fender | yes, well there are 3 cases in the PB, and you showed me 1 GotoIF. So they cannot be considered equivalent just yet |
| 00:59.46 | [TK]D-Fender | yes, in 1 little bit it can lok a little more natural, but you pay a price for it elsewhere |
| 01:00.05 | pdtpatrick | which is compiling back to basic dialplan ? |
| 01:00.15 | [TK]D-Fender | pdtpatrick: Have you looked as I suggested? |
| 01:00.20 | pdtpatrick | checking now |
| 01:01.09 | *** join/#asterisk coppice (~chatzilla@116.92.17.112) |
| 01:01.30 | [TK]D-Fender | pdtpatrick: Feel free to pastebin that as well. good for reference |
| 01:05.35 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 01:11.46 | pdtpatrick | ugh ael already giving me problems |
| 01:11.54 | pdtpatrick | [Sep 26 18:11:29] ERROR[6548]: pbx_ael.c:197 pbx_load_module: Sorry, but 2 syntax errors and 0 semantic errors were detected. It doesn't make sense to compile. |
| 01:17.09 | [TK]D-Fender | JabberSend(asterisk-xmpp,bob@jabber.org,(Calling cellphone...); <- mismatched parenthesis |
| 01:17.18 | [TK]D-Fender | x2 |
| 01:17.27 | [TK]D-Fender | as far as I could see right off the bat... |
| 01:18.22 | [TK]D-Fender | Those are the clearer syntax ones. your Dial's also look wrong as far as specifying peers would appear (and your use of ${EXTEN} i suspect may not be what you were thinking of it for either) |
| 01:18.33 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 01:20.24 | pdtpatrick | fixed that .. now it is complaining about this |
| 01:20.38 | pdtpatrick | [Sep 26 18:19:40] ERROR[6636]: ael.y:840 ael_yyerror: ==== File: /etc/asterisk/extensions.ael, Line 4, Cols: 9-19: Error: syntax error, unexpected word, expecting ';' or '=' |
| 01:21.58 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 01:23.26 | sawgood | I can start Asterisk with option(s) to store all CLI output to a 'txt file (and this does serve as a very valuable method for some types of troubleshooting, but I would like know if someone can offer a suggestion for this ...(follows) |
| 01:23.53 | [TK]D-Fender | pdtpatrick: You might want to pay attention to your variable and function references.... |
| 01:24.08 | sawgood | Once Asterisk has started (and you are at the CLI), is there a tip you can offer to where I can start something like (sip set debug on), but have the output go to a txt file instead of the screen? |
| 01:24.16 | pdtpatrick | will check again and update |
| 01:24.49 | sawgood | So, I can take the 'text' file, and open it with a SIP troubleshooting tool? |
| 01:25.10 | p3nguin | How do you connect to the asterisk computer? |
| 01:25.15 | sawgood | SSH only |
| 01:25.23 | p3nguin | With what ssh client? |
| 01:25.34 | sawgood | right now, I am running two SSH clients for various reasons |
| 01:25.49 | p3nguin | With what ssh clients? |
| 01:25.50 | sawgood | well, I use putty from a Win X box and standard SSH from a Debian box |
| 01:26.00 | p3nguin | If you use PuTTY, it does logging to file. Use that. |
| 01:26.18 | sawgood | that is one choice ... thank you |
| 01:26.35 | sawgood | at least with that option, the file is LOCAL to my client side |
| 01:31.09 | [TK]D-Fender | keep in mind how much random taffic goes on with qualify packets alone. Your logging will be enormous very quickly |
| 01:31.37 | [TK]D-Fender | And if you are doing this remote from your server LAN that is also added traffic to keep in mind (if you're really tight on things) |
| 01:33.31 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
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| 01:37.18 | sawgood | I decided to use a network TAP on another box with a Wireshark approach |
| 01:37.38 | sawgood | [TK]D-Fender: thank you so much as well |
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| 04:50.12 | Micc | where can I find information on what ports to open in iptables for tcp transport? |
| 04:57.13 | ChannelZ | tcp SIP is 5060 AFAIK |
| 04:57.31 | ChannelZ | but RTP is still UDP |
| 04:59.17 | Micc | is there any way to do rtp over tcp? |
| 05:00.37 | ChannelZ | no TCP is shitty for realtime media |
| 05:01.35 | Micc | that is true. |
| 05:03.02 | kaldemar | the SIP port is what you configure it to be in sip.conf |
| 05:04.21 | kaldemar | by default it is 5060 |
| 05:07.33 | Micc | if its just the sip port, thats easy. for some reason I thought it was rtp too. |
| 05:07.55 | Micc | I'm hoping the tcp connection open to the server will help this particular router that is doing funny things. |
| 05:24.34 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
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| 06:11.33 | *** join/#asterisk din3sh (29d4d142@gateway/web/freenode/ip.41.212.209.66) |
| 06:11.54 | din3sh | Hi all |
| 06:12.49 | din3sh | Can anyone confirm if the CDR issue https://issues.asterisk.org/view.php?id=11849 has been resolved in 1.8.x or the only way out is to use CEL? |
| 06:31.46 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
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| 06:42.04 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 06:43.18 | *** join/#asterisk jkroon (~jkroon@41.49.6.134) |
| 06:45.24 | jkroon | has anyone ever seen "SIP/2.0 200 Auth Failed" in response to a REGISTER packet? |
| 06:45.52 | jkroon | nearly burst ot laughing and crying at the same time. |
| 06:46.13 | ChannelZ | NEVAR! |
| 06:46.35 | jkroon | real live, curtessy of mweb in ZA. |
| 06:46.53 | jkroon | Sip EXpress running on freebsd. |
| 06:49.39 | jkroon | apparently it works with a soft phone, so just wondering how I can work around it. |
| 06:50.44 | ChannelZ | oh you're really asking? |
| 06:53.19 | jkroon | i'm afraid so. |
| 06:53.44 | ChannelZ | oh.. well the simple answer is fix your peer/credentials |
| 06:54.47 | jkroon | uhrm, secret is correct ... as is the username (as far as I can tell), so I just don't know what more to look at ... |
| 06:55.06 | ChannelZ | What does the console say? |
| 06:56.06 | ChannelZ | its probably not matching the right peer |
| 06:56.18 | jkroon | registered. it is getting a SIP/2.0 200 response after all./ |
| 06:57.14 | jkroon | i'll pastebin the registration exchange, perhaps you can spot something from that. |
| 06:57.55 | ChannelZ | well if it's working..... |
| 06:58.40 | jkroon | that's the point. provider systems says not registered :p |
| 06:59.37 | ChannelZ | who said "200 Auth failed"? |
| 07:00.16 | jkroon | http://pastebin.com/vbjZy1g1 |
| 07:00.20 | jkroon | the provider. |
| 07:00.57 | jkroon | asterisk in that exchange is the client. |
| 07:01.53 | ChannelZ | ohhhh |
| 07:03.50 | wdoekes2 | jkroon: you do know that you're advertising .35 while it sees .34 ? |
| 07:04.35 | jkroon | wdoekes2, no i did not. |
| 07:05.09 | jkroon | tries to see if he can get that fixed and if it helps. |
| 07:07.00 | jkroon | fixed with externip, no change. |
| 07:08.46 | ChannelZ | well they don't appear to like your auth, for reasons I don't know |
| 07:10.06 | jkroon | then why don't they respond with a 403 ?!? guess I'll have to ask them that. |
| 07:10.49 | din3sh | @jkroon works on softphone but not hard ones? |
| 07:11.16 | din3sh | is the SIP port used same for both softphone and hard ones? |
| 07:11.28 | din3sh | might be a firewall issue |
| 07:11.29 | kaldemar | jkroon: grab a registration with a soft phone and compare it to the asterisk one. |
| 07:11.43 | ChannelZ | hmm true are these all behind the same NAT? |
| 07:11.53 | jkroon | yes |
| 07:12.06 | jkroon | din3sh, pretty much any "IP Phone" according to MWeb. |
| 07:12.18 | jkroon | so soft phone was a misnomer on my side. |
| 07:12.50 | jkroon | din3sh, i don't think it's a firewall issue, at least not at the registration stage, port 5060 traffic is passing through and I am getting responses. |
| 07:13.09 | jkroon | I did alter it so that I advertize .34 as my public IP using externip instead of externhost ... |
| 07:14.16 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:16.12 | kaldemar | jkroon: have you tried it with a phone? |
| 07:17.18 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 07:17.26 | jkroon | kaldemar, going to get that set up in a second. |
| 07:17.48 | din3sh | are these all behind the same NAT? |
| 07:18.16 | jkroon | the phone won't be no |
| 07:18.17 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:18.35 | jkroon | unfortunately. |
| 07:18.57 | jkroon | ok, where can I find the rfc on calculating the md5 hashes for the response to the auth challenge? |
| 07:19.34 | din3sh | i think ure going too far there, the answer might be simpler |
| 07:20.08 | din3sh | compare the softphone and ip phone and see whats not constant |
| 07:20.10 | jkroon | please explain ? |
| 07:20.25 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f49e:600a:82d3:b620) |
| 07:20.50 | jkroon | the only access i've got behind that NAT is the asterisk server ... so can't set up a phone behind that particular nat. |
| 07:22.02 | din3sh | the router in front of the asterisk server allows all ports? |
| 07:24.17 | kaldemar | jkroon: http://en.wikipedia.org/wiki/Digest_access_authentication |
| 07:27.06 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:27.56 | din3sh | Can anyone confirm if the CDR issue https://issues.asterisk.org/view.php?id=11849 has been resolved in 1.8.x or the only way out is to use CEL? |
| 07:32.05 | kaldemar | din3sh: IMO mnicholson quite clearly states that the issue is closed as won't fix. i wouldn't expect it to be fixed in any version until that issue says so. https://issues.asterisk.org/jira/browse/11849 is a better place to follow that since mantis is not used anymore. |
| 07:33.03 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 07:33.18 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-194-204-114.zone8.bethere.co.uk) |
| 07:33.41 | jkroon | kaldemar, thanks. |
| 07:34.39 | jkroon | kaldemar, ChannelZ, din3sh, the SIP phone registration exchange: http://pastebin.com/dbUSuNX5 |
| 07:35.58 | din3sh | :S @kaldemar, i've tested 1.8.7.0 and even 10, the CDR missing on transfer issue doesnt seem to have bee fixed :/ |
| 07:36.05 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
| 07:38.26 | jkroon | din3sh, no it's not fixed. and won't be. the channel gets transferred so it really depends on which channel the cdr was initiated on. |
| 07:38.45 | jkroon | two workarounds, work via Local/??/n channels and generate CDRs on those, or use CEL |
| 07:39.23 | wdoekes2 | jkroon: do they have an IP-white-/blacklist? |
| 07:39.32 | din3sh | @jkroon Contact: <sip:27877008071@192.168.47.12:2048; |
| 07:39.40 | din3sh | the port isnt supposed to be 5060? |
| 07:39.41 | jkroon | wdoekes2, i'll ask. |
| 07:39.44 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:39.47 | jkroon | that's the phone side. |
| 07:40.21 | din3sh | phone side 2048 nat to 5060 asterisk side? |
| 07:40.25 | jkroon | it seems they do the sane thing with NAT=auto and detect people that's behind NAT. |
| 07:40.34 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:40.36 | schmidts | good morning |
| 07:40.42 | wdoekes2 | (and you could try removing externip= altogether.. seeing that the phone sends rfc1918) |
| 07:41.50 | jkroon | wdoekes2, still results in SIP/2.0 200 Auth Failed |
| 07:42.07 | jkroon | without externip. |
| 07:43.05 | wdoekes2 | yea.. a blacklist is more plausible (e.g. when the auth matching does say 200, but they fail afterwards because of the unwanted IP) |
| 07:43.20 | din3sh | can u test without the nat configuration in between? then if the phone does register, u'd at least know that the prob is with ur nat config |
| 07:43.31 | jkroon | bp @ http://pastebin.com/EvJJFxtn |
| 07:43.41 | irroot | jkroon still having the fd problem |
| 07:43.45 | jkroon | i've been trying to eliminate that NAT with almost every second breath last week. |
| 07:43.52 | jkroon | irroot, yes i am. even on 1.8.7.09-rc2 |
| 07:43.57 | jkroon | irroot, yes i am. even on 1.8.7.0-rc2 even |
| 07:44.19 | wdoekes2 | jkroon: that last one is odd: why don't you get a received= this time? |
| 07:44.42 | jkroon | wdoekes2, in one of my more unhelpful responses simply because i don't no: green |
| 07:44.59 | irroot | the fromdomain setting ?? |
| 07:45.25 | jkroon | wdoekes2, the SNOM300 passes Supported: gruu ?? |
| 07:45.26 | irroot | see the realm= maybe need it ?? |
| 07:45.38 | jkroon | not sure if that could cause the different effect? |
| 07:45.40 | wdoekes2 | no.. that's your realm |
| 07:45.59 | jkroon | fromdomain is currently set to the IP |
| 07:46.14 | jkroon | tried with a horde of their various system names, all with the same result. |
| 07:46.25 | jkroon | also, the SNOM phone works with the fromdomain equal to the IP. |
| 07:47.15 | irroot | not fromdomain realm |
| 07:47.29 | irroot | maybe need the realm set properly |
| 07:47.32 | din3sh | does realm on phone config match realm on asterisk> |
| 07:49.03 | din3sh | set realm=jkroon on both |
| 07:51.58 | wdoekes2 | I cannot see any reason at all why received= is gone in that last dump. I think you'll have to contact them because something seems messed up in their rules |
| 07:52.35 | wdoekes2 | there's one thing left I see: the name-addr "Jaco Kroon" which you don't get in the asterisk trace |
| 07:55.20 | irroot | wdoekes2 jkroon we know the morons who are running this :P |
| 07:56.47 | jkroon | wdoekes2, no blacklisting going on. |
| 07:57.09 | jkroon | irroot, MWeb. |
| 07:57.17 | jkroon | on the phone with them now - any questions? |
| 07:57.18 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:57.39 | irroot | tell them i send my regards ... of course im been polite |
| 08:02.45 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 08:04.10 | jkroon | done. |
| 08:05.42 | jkroon | ok, other ideas? |
| 08:06.51 | jkroon | Via: SIP/2.0/UDP 192.168.47.12:2048;branch=z9hG4bK-1ht4rz27a7be;rport <-- phone sends rport on the register request - how can I get asterisk to follow suit? |
| 08:07.58 | wdoekes2 | so does asterisk (nat=something-with-yes) (see your first http://pastebin.com/vbjZy1g1 ) |
| 08:08.53 | wdoekes2 | and there went africa |
| 08:10.52 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 08:18.29 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 08:20.15 | irroot | . |
| 08:20.59 | *** join/#asterisk IsUp (5b8e8e8f@gateway/web/freenode/ip.91.142.142.143) |
| 08:21.10 | IsUp | hello |
| 08:21.21 | IsUp | is that possible to remove '+' sign from CallerID in dialplan? |
| 08:22.31 | kaldemar | IsUp: of course. where is it in the caller id? |
| 08:23.43 | IsUp | kaldemar: ${CALLERID(num)} returns me: +165452200, i want to remove that '+' sign. but i telco sending me calls, sometimes they are not putting '+' on Caller IDs. so CUT function is not working correctly if theres no '+' |
| 08:24.30 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 08:24.52 | irroot | IsUp use gotoif($[${VAR:1:1} = +] .... |
| 08:24.54 | IsUp | kaldemar: i think its resolved now, ive tried 1-2 in CUT function |
| 08:25.14 | IsUp | ${CUT(${CALLERID(num)},'+',1-2)} |
| 08:26.29 | kaldemar | IsUp: Execif($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num)=${CALLERID(num):1})) |
| 08:26.40 | kaldemar | something like that. |
| 08:26.41 | irroot | kaldemar +1 |
| 08:26.42 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:28.18 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:29.30 | IsUp | kaldemar: is that wrong to use CUT? because that works for me now: Set(callerfilter=${CUT(${CALLERID(num)},'+',1-2)}) |
| 08:31.21 | kaldemar | there is more than one way to do it. here's one more: Set(CALLERID(num)=${REPLACE(CALLERID(num),+)}) |
| 08:33.37 | kaldemar | i'd like execif or replace better because there really are no fields but a single character. but you can be the judge of which you like the most. |
| 08:34.10 | IsUp | kaldemar: I dont have REPLACE function, and i dont know why. i am running 1.4. i cant see it in 'show functions' |
| 08:38.26 | kaldemar | REPLACE is 1.8 only. |
| 08:39.39 | IsUp | ah okay |
| 08:39.42 | IsUp | thank you so much |
| 08:40.30 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 08:54.05 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 08:57.53 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 09:00.25 | IsUp | kaldemar: it says " execif_exec: Invalid Syntax." |
| 09:00.53 | kaldemar | what exactly? my example? |
| 09:01.01 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 09:02.45 | IsUp | kaldemar: yes, probaby ? was wrong, ive changed it "," also Set( was wrong, i set Set,params and its working i think :p |
| 09:03.27 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:03.57 | IsUp | kaldemar: thanks for the idea, CUT didnt worked well |
| 09:04.50 | Lipsum | When working with XML objects (particularly CiscoIPPhoneInput) on SPA504, is it somehow possible to enable extended ascii support for the input fields? The SPA525 has support for them by default it appears. |
| 09:05.28 | *** join/#asterisk linuxplatform (~centoslin@88.87.48.115) |
| 09:09.27 | kaldemar | IsUp: the execif i gave earlier works fine on 1.8. |
| 09:09.33 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 09:09.36 | *** join/#asterisk dwayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 09:12.57 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 09:17.53 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 09:22.02 | Faustov | for the purpose of some filtering I'm looking at CIDs coming in to my asterisk box, some calls from Madrid come as "nian<number>" for some reason - anyone got any idea why this might be happening? |
| 09:23.13 | kaldemar | Faustov: the party that sends you the calls might. |
| 09:28.48 | *** join/#asterisk Tribbers (~joey@host217-37-142-238.in-addr.btopenworld.com) |
| 09:29.37 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 09:39.03 | *** join/#asterisk alfadir (~alfadir@79.141.248.181) |
| 09:41.02 | alfadir | hi, connecting to a sip testnr with linphone. seems to work.. but no sound. I get message: bandwidth usage: audio=[d=0.0,u=81.4] video=[d=0.0,u=0.0] kbit/sec in the debug. any ideas? no audio downstream ? is there a better client ? other problems ? |
| 09:41.44 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 09:43.57 | Faustov | kaldemar: sure, I'm trying to find out if this is a local misconfiguration or some standard, however google doesn't tell me much |
| 09:44.38 | *** join/#asterisk black187 (5d67162a@gateway/web/freenode/ip.93.103.22.42) |
| 09:46.20 | black187 | Hello guys. Does anybody know what exactly asterisk needs for performance - CPU, RAM? We are trying to assemble a hardware for around 4000 users - do we use 64 bit proc.? Do we use Opteron?... |
| 09:46.41 | black187 | The hardware would be for testing purposes -> stress testing. |
| 09:48.06 | black187 | And regarding stress testing - we use sipp, but is there any other tool like sipp, but that it can make a call (RTP) and not play some file from disk? |
| 09:48.55 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 09:54.15 | kaldemar | black187: can sipp not make a call with RTP? |
| 09:55.11 | black187 | We tried, but it didn't work -> maybe a dialplan issue. |
| 09:58.22 | Tribbers | Hi Asterisk has been configured to start on boot and when my server boots up I can make calls and everything works expect for voicemail. Voicemail only starts working when i call asterisk. Any reason why the voicemail wouldnt work even though asterisk is running? |
| 09:59.47 | IsUp | Tribbers: Do you have any output? logs? |
| 10:01.24 | Tribbers | I have looked at the logs and there are no obvious errors. Postfix is definitely started. Cannot really check if the CLI output is calling the voicemail, as soon as i go into the CLI it starts working |
| 10:04.07 | *** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178) |
| 10:04.36 | SteveWilliams | Hi All! Good Morning! |
| 10:07.20 | SteveWilliams | Please Help! When I type odbc show in the asterisk console, it says that no such function exists. How do I install / enable the odbc function? I am using Asterisk 1.4 |
| 10:08.19 | SteveWilliams | Is there a similar function available for my version |
| 10:08.24 | irroot | SteveWilliams you need to build asterisk with odbc drivers unixodbc also remember the header files are needed if you using some package |
| 10:09.16 | SteveWilliams | irroot:Okay. Thanks! |
| 10:20.08 | Tribbers | IsUp: I apoligise I do have errors. It appears to be that getting asterisk to start up on boot simply just allows phone calls to be made but teh dialpan is ignored as it just rings a panic group, voicemails don't work, voice recordings dont work. As soon as I run asterisk, everything runs as it should. Do I not understand what make config actually does? |
| 10:21.38 | Tribbers | Or rather than ringing a panic group as it is definitely not touching the dialplan it just rings every registered phone |
| 10:26.32 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 10:26.55 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 10:31.42 | kaldemar | Tribbers: make config just installs an init script. |
| 10:37.01 | Tribbers | kaldemar: so does this not start asterisk then, sorry I am new to this game any help would be greatly appreciated. I may just need to go and read up some more |
| 10:49.31 | *** join/#asterisk jkroon (~jkroon@dsl-241-237-66.telkomadsl.co.za) |
| 11:01.07 | IsUp | Tribbers: how you start asterisk? any parameters? |
| 11:04.04 | _naomi | LEAVE |
| 11:04.09 | _naomi | sorry was trying to leave |
| 11:04.29 | *** part/#asterisk _naomi (~naomi@79.135.102.10) |
| 11:04.47 | Tribbers | IsUp: I have run the make config and I also ran chkconfig asterisk on, which i am under the presumption will also start asterisk on boot. However at the moment to get asterisk working i simply type "asterisk" and everything starts working as it should. |
| 11:07.07 | Tribbers | If it is any help when i rebooted i checked the status of asterisk and it says it is running. But again it is not actually functional until I type "asterisk". |
| 11:10.34 | IsUp | Tribbers: when you rebooted, just do 'asterisk -r' and then see if you are able to connect to console. if you can, do 'module show like app_voicemail.so' |
| 11:11.09 | jkroon | irroot, what do you think the chances are that mweb has a list of allowed user agent strings? |
| 11:11.21 | jkroon | ie, to combat things like sipwich |
| 11:11.59 | jkroon | wdoekes2, ? |
| 11:12.17 | wdoekes2 | 60% ;P |
| 11:12.27 | irroot | jkroon not sure i doubt they smart enough but they may be dumb enough to try and fuck it up |
| 11:12.44 | jkroon | proceeds to fake the user agent to what is being used by a SNOM300 |
| 11:13.14 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 11:13.59 | jkroon | nope. |
| 11:14.05 | black187 | Does anybody know what exactly asterisk needs for performance - CPU, RAM? We are trying to assemble a hardware for around 4000 users - do we use 64 bit proc.? Do we use Opteron?.. |
| 11:14.06 | jkroon | perhaps something related to the Contact: header ?!? |
| 11:14.23 | as001 | Hello, I am using Asterisk 1.6.2.20, my agents are in queues and Agent/XXX is paused, when I rewrite configuration and do reload on CLI paused agent gets call despite he is paused. How can i prevent that ? |
| 11:14.33 | jkroon | black187, i've got around 500 users on about 200MB of RAM, CPU depends on call concurrency. |
| 11:14.52 | jkroon | 2.5GHz (Xeon processor) per approximately 50 concurrent calls should be OK. |
| 11:15.02 | jkroon | don't underestimate your disks if you're doing call recording. |
| 11:15.16 | Tribbers | IsUp: I dont believe it is just voicemail now, you may not have read my message earlier. However I have just done what you said and the message is "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)". yet making calls still work so asterisk is up in some way. |
| 11:15.46 | black187 | @jkroon: Ok thanks - the procesor is mainly responsible for concurrent calls, RAM for SIP registration? |
| 11:15.54 | alfadir | hi, trying to call a testnr from nat to server. sending ok, but receiving registerd as zero. tried 2 clients on different os.. a firewall problem ? |
| 11:16.11 | alfadir | connection, commands ok |
| 11:16.21 | alfadir | just no sound from the other end.. |
| 11:16.30 | jkroon | black187, something to that effect yes. there is per-call memory constructs too, obviously. |
| 11:17.24 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 11:17.51 | IsUp | Tribbers: are you running your asterisk as root? |
| 11:19.08 | black187 | @jkroon: Ok thanks... |
| 11:19.22 | Tribbers | IsUp: yes, I am the only one working on the server and I have not not added any other users :D |
| 11:28.16 | IsUp | Tribbers: its strange, i cant understand how you are making calls when you cant connect to console |
| 11:29.17 | as001 | Hi why are my agents receive calls from queue after reload when they are paused ? |
| 11:31.00 | Tribbers | IsUp: I know I just need to find out why asterisk is starting up properly. Thanks for your time anyway really appreciate it. Will carry on searching see if I can find anything. |
| 11:31.35 | IsUp | Tribbers: no problem, i am kinda busy and sorry for delay. you should check your logs. good luck |
| 11:37.15 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 11:39.30 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 11:41.05 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 11:50.47 | *** part/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 11:51.18 | *** join/#asterisk coppice (~chatzilla@m121-202-106-154.smartone-vodafone.com) |
| 11:52.21 | *** join/#asterisk russellb (~russellb@asterisk/contributor-and-cool-guy/russellb) |
| 11:52.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 11:54.11 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 11:58.55 | *** join/#asterisk Verzuz (~Verzu@79.189.114.202) |
| 12:00.27 | depressed | hello |
| 12:00.44 | *** part/#asterisk atheos (~atheos@208.119.68.16) |
| 12:00.55 | ollii | hi |
| 12:00.55 | *** join/#asterisk atheos (~atheos@208.119.68.16) |
| 12:00.55 | *** part/#asterisk atheos (~atheos@208.119.68.16) |
| 12:01.12 | Verzuz | hi, does my voip phone need to support reinvite to properly work with directmedia=yes (or in older version canreinvite=) variable? |
| 12:01.45 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:04.35 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:05.45 | kaldemar | Verzuz: properly? it needs to support pre-session re-invites. |
| 12:07.30 | as001 | Do you know does reload of asterisk do something wrong with paused Queue members so they can receive calls despite they have been paused before reload ? |
| 12:08.27 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:08.37 | Verzuz | kaldemar, properly - im trying to set it up on some experimental simple-config asterisk, but got much problems with it, i will check pre-session then, thanks |
| 12:14.47 | jkroon | i am going to kill one of my tech staff! |
| 12:15.05 | jkroon | right, ok, got it up to one-way voice. which is an issue with NAT itself. |
| 12:15.27 | irroot | jkroon you tube it ... can we help/watch |
| 12:15.57 | jkroon | i've now got to go explain to a client why it took me more than a week to track an incorrect password. |
| 12:16.23 | jkroon | I will blame it on the fact that MWeb gives me SIP/2.0 200 even on incorrect auth details, thus indicating to asterisk that authentication was successful. |
| 12:16.52 | jkroon | ok, is there magic voodoo I can feed the remote peer to request it only start sending rtp once it starts receiving from me? |
| 12:20.24 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 12:23.55 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 12:39.33 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 12:39.37 | jacc0 | hi all |
| 12:44.25 | *** join/#asterisk coppice (~chatzilla@116.92.17.112) |
| 12:48.10 | Verzuz | kaldemar, what do you mean by pre-session re-invite? it's rly hard tofind anything about it on the net... |
| 12:48.55 | irroot | jkroon they need to fix there proxy then |
| 12:51.09 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 12:51.10 | otwieracz | Hello. |
| 12:51.23 | otwieracz | Is available in Asterisk to automatically add some contacts to user roster? |
| 12:52.28 | jkroon | irroot, or i need to get gamco to get IS to get some udp port forwarding done :p |
| 12:52.30 | *** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178) |
| 12:53.26 | [TK]D-Fender | otwieracz, Asterisk doesn't manage anything conceptually like "contacts" |
| 12:56.23 | atan | Anyone know if the Plantronics CS361N works well with the DA40/DA55/DA60? |
| 12:57.40 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:57.59 | puzzled | hi |
| 12:58.38 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:58.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:09.41 | jkroon | weird. |
| 13:09.52 | jkroon | normal call works, Echo() does not. |
| 13:11.04 | [TK]D-Fender | jkroon, show us |
| 13:12.38 | jkroon | [TK]D-Fender, what do you need to see? In the Echo() case I'm not receiving rtp from the provider, so not very unexpected to not receive anything back ... |
| 13:13.04 | jkroon | not too phazed about that particular issue in the bigger scheme of things to be honest. |
| 13:15.22 | [TK]D-Fender | jkroon, PB the call |
| 13:15.48 | kaldemar | Verzuz: by pre-session i mean that the call (SIP session) is not fully set up yet. |
| 13:16.49 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:16.55 | jkroon | [TK]D-Fender, busy setting it up for a capture. |
| 13:24.34 | *** join/#asterisk serafie (~erin@nat/digium/x-nsstxifarpwovisn) |
| 13:27.11 | Katty | drags in |
| 13:27.53 | irroot | gives katty hugs and big wimpy mega coffee |
| 13:29.35 | WIMPy | is not a coffee |
| 13:29.53 | *** part/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 13:30.03 | Katty | hugs irroot |
| 13:30.09 | Katty | hugs WIMPy |
| 13:30.16 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 13:30.22 | Katty | wimpy needs 100mg caffeine STAT |
| 13:30.49 | WIMPy | That's not going to wake me up. |
| 13:30.50 | irroot | Katty lol got you wimpy is a chain here that is known for coffee something like a wendys/burger king |
| 13:31.37 | Katty | i think we used to have a wimps here long ago |
| 13:31.40 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:31.44 | Katty | long before i was born tho |
| 13:33.28 | Katty | http://www.capecentralhigh.com/wp-content/uploads/2010/02/Wimpys-motorcycle-wreck-1967-.jpg <- 1967 |
| 13:37.12 | irroot | intresting |
| 13:37.17 | Katty | 17 years before i was born tho |
| 13:38.29 | tzanger | there's a Wimpy's diner just a few km from me |
| 13:38.33 | tzanger | decent breakfast, although slow |
| 13:38.58 | irroot | http://www.wimpy.co.za/index.asp |
| 13:39.09 | coppice | Wimpy (named after the character in Popeye) was *the* burger bar long before McD were known at all. Anyone know what happened to them |
| 13:39.12 | Katty | must have been pretty popular |
| 13:39.29 | Katty | the home of popeye isn't too far from here |
| 13:39.48 | coppice | you do realise he is fictional, don't you? |
| 13:40.10 | Katty | bout an hour from here in Chester IL |
| 13:42.17 | *** join/#asterisk Vilius_Invade (~Vilius_In@178.78.119.76) |
| 13:42.33 | tzanger | coppice: nonsense, it is totally possible to squeeze a tin can and have the spinach leap up in a perfect arc into your mouth |
| 13:43.04 | miztic | yeah thats the part i didn't buy either :) |
| 13:43.07 | coppice | tzanger: it probably is, but Popeye is still fictional |
| 13:43.58 | WIMPy | has once had fish&chips at a Wimpy's in London. |
| 13:44.26 | Katty | would you believe i don't like fish n chips? |
| 13:44.46 | coppice | Wimpy's restaurants in England used to be quite expensive |
| 13:45.02 | irroot | coppice tzanger popeye was a sailor you try tell a sailor they not real and see how it works out |
| 13:45.52 | [TK]D-Fender | senses great Venn failure... |
| 13:48.17 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:49.19 | coppice | irroot: Popeye was a "sailor man", and not the don't ask don't tell kind. most unreal |
| 13:49.52 | irroot | coppice the YMCA kind ?? |
| 13:50.17 | coppice | The cop. The Indian brave. The Popeye |
| 13:52.00 | *** join/#asterisk kpettit (~kpettit@99-116-144-138.lightspeed.hstntx.sbcglobal.net) |
| 13:52.11 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 13:53.40 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:54.22 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:54.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:57.20 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-189.dsl.stlsmo.sbcglobal.net) |
| 13:58.16 | LemensTS | I installed oslec into dahdi, and i was wondering in /etc/dahdi/system.conf do I need to change it to echocanceller=oslec ? I hadn't seen it say anything about doing that |
| 13:59.25 | *** join/#asterisk micols (~0x2AA7F64@rlogin.dk) |
| 14:02.34 | [TK]D-Fender | LemensTS, You do |
| 14:06.04 | *** join/#asterisk master_of_master (~master_of@p57B5452F.dip.t-dialin.net) |
| 14:12.19 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 14:13.23 | LemensTS | TKD-Fender: figured so, thanks. ps nice to see ya back here |
| 14:13.34 | mirko_brankovic | Does anyone know why i get 484 Address Incomplete back from 'IP' when using Dial(Local/xxxx) to extension that exists |
| 14:14.20 | mirko_brankovic | using 1.8.5.0. version on 2 servers, on test one it works, but on live one it doesn't |
| 14:15.01 | [TK]D-Fender | mirko_brankovic, Pastebin the call with SIP debug enabled along with your dialplan. |
| 14:15.19 | [TK]D-Fender | mirko_brankovic, And we'll be better able to show you where & why |
| 14:16.26 | mirko_brankovic | i know :) but i need some time to get that log. I'll try to get it today, if not, thx for reply :) |
| 14:17.41 | p3nguin | mirko_brankovic: Is 'xxxx' a valid 'exten@context'? |
| 14:18.09 | [TK]D-Fender | mirko_brankovic, It certainly helps when we can see what's going on. Do come back with all of that backup soon.... |
| 14:18.48 | p3nguin | By valid, I mean the context exists and the exten is within it. |
| 14:18.50 | wdoekes2 | ~ask |
| 14:18.50 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:20.04 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-251-132.w81-51.abo.wanadoo.fr) |
| 14:21.28 | merlin8282 | Hi. I have the problem that GoSub(intern-${DIALSTATUS},1) does not work anymore, after upgrading from 1.6 to 1.8. It should go to _intern-.,1,Goto(intern-NOANSWER,1) but it doesn't. Is this a known issue, or is it me ? |
| 14:21.57 | merlin8282 | I also saw following error : app_stack.c:420 gosub_exec: Attempt to reach a non-existent destination for gosub: (Context:default, Extension:intern-CHANUNAVAIL, Priority:1) |
| 14:22.39 | merlin8282 | The [intern] context is included in [default] (where the calls come in). |
| 14:23.25 | mirko_brankovic | p3nguin yes |
| 14:23.46 | merlin8282 | I also have [compat] pbx_realtime=1.6 res_agi=1.6 app_set=1.6 in asterisk.conf |
| 14:23.53 | p3nguin | merlin8282: That's a terrible design. |
| 14:23.56 | mirko_brankovic | p3nguin, yes, it exists |
| 14:24.02 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-163-189.dsl.stlsmo.sbcglobal.net) |
| 14:24.30 | p3nguin | merlin8282: Regardless of design fault, it says the DIALSTATUS is CHANUNAVAIL. It will never match NOANSWER. |
| 14:25.03 | [TK]D-Fender | mirko_brankovic, pastebin your dialplan at least. Perhaps everything will be evident in there... |
| 14:25.45 | merlin8282 | p3nguin: ok, so what do you recommend ? I simply took the example from voip-info and modified it to my needs. Also, it should match "_intern-.", no ? |
| 14:26.26 | p3nguin | merlin8282: I don't know about matching that pattern, so I wouldn't use it. I would define extensions for intern-CHANUNAVAIL and intern-NOANSWER. |
| 14:27.04 | p3nguin | I would also create intern-BUSY and intern-CONGESTION. |
| 14:27.19 | [TK]D-Fender | in your use of "intern", "n" is a reserved letter for a numeric match. |
| 14:27.22 | irroot | p3nguin will be testing the mp3 bug soon have not had time but seems to work |
| 14:27.30 | [TK]D-Fender | You have to be careful throwing alpha chars around like that. |
| 14:28.06 | p3nguin | irroot: You don't have the problem I've encountered? |
| 14:28.11 | [TK]D-Fender | So a much safer and better alternative is to use priority labels instead. |
| 14:28.22 | irroot | no sorry but need to look at it more in depth |
| 14:28.26 | p3nguin | Or just use the extensions like I've defined. |
| 14:28.40 | p3nguin | They aren't patterns, so n doesn't match anything but n. |
| 14:28.55 | p3nguin | His way breaks because of patterns. |
| 14:30.30 | mirko_brankovic | [TK]D-Fender, p3nguin, http://pastebin.com/t9YpdUUB |
| 14:31.41 | [TK]D-Fender | mirko_brankovic, we can't see what {QueueToEnter} will evaluate to... |
| 14:32.15 | p3nguin | merlin8282: http://pastebin.com/B7PZL6kW |
| 14:32.39 | mirko_brankovic | [TK]D-Fender it will be 4XXX any digit |
| 14:32.54 | [TK]D-Fender | mirko_brankovic, I also don't see why you didn't make that a macro/gosub instead of a sub-dialed |
| 14:32.58 | mirko_brankovic | depends from service that queue uses |
| 14:32.59 | merlin8282 | p3nguin: ok, I understand. I'll do it like this ;) |
| 14:33.54 | *** join/#asterisk Fritz09 (~Adium@pop1-73.catv.wtnet.de) |
| 14:34.16 | mirko_brankovic | [TK]D-Fender had to be dial cause of g option, so that callee can go back to AEL code |
| 14:35.29 | [TK]D-Fender | mirko_brankovic, if you just used a macro/gosub you wouldn't need to... it'd fall back anyway |
| 14:35.44 | [TK]D-Fender | <mirko_brankovic> [TK]D-Fender it will be 4XXX any digit <- BTW, this is why it fails |
| 14:36.10 | [TK]D-Fender | mirko_brankovic, exten => _Q4XXX.,1,PauseQueueMember(,Agent/${CALLERID(number)}) <-- this requires 4XXX and at least one more character |
| 14:36.34 | [TK]D-Fender | mirko_brankovic, "." means 1 or more. Yuo put that at the end of something that was already 4 long, meaning 5+ |
| 14:37.24 | mirko_brankovic | [TK]D-Fender aha so it won't accept 4 digits |
| 14:37.25 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 14:37.27 | merlin8282 | [TK]D-Fender: ah, so patterns are case insensitive, e.g. 'n' is equal to 'N' ? |
| 14:37.39 | [TK]D-Fender | merlinyes |
| 14:37.46 | [TK]D-Fender | mirko_brankovic, Correct |
| 14:38.02 | mirko_brankovic | [TK]D-Fender thank you very much :) |
| 14:38.10 | [TK]D-Fender | mirko_brankovic, You're welcome |
| 14:38.54 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 14:41.49 | merlin8282 | Ok, now it works (I changed "intern-" to "dialstatus-"). |
| 14:42.24 | *** join/#asterisk charley (boise@epicboise.com) |
| 14:42.40 | Qwell | way to work around the problem |
| 14:44.17 | [TK]D-Fender | merlinI highly recommend you switch to using a basic exten and jumping based on named priority. |
| 14:44.28 | [TK]D-Fender | merlin8282, ^ |
| 14:44.52 | merlin8282 | mmm, ok. I understand |
| 14:44.55 | [TK]D-Fender | Even within the same pattern |
| 14:45.49 | p3nguin | For some reason, I don't think you do. |
| 14:45.57 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 14:46.02 | ollii | maybe he needs an example |
| 14:46.09 | Qwell | ollii: he was given like 3 |
| 14:46.35 | merlin8282 | so instead of having a pattern like "_dialstatus-.", it is better to use "dialstatus-CHANUNAVAIL", "dialstatus-BUSY", etc. |
| 14:46.43 | p3nguin | I gave him one method, and he only had to copy/paste it. |
| 14:46.44 | merlin8282 | ok, ok |
| 14:46.55 | *** join/#asterisk pdtpatrick1 (~pdtpdt@ip72-211-209-214.oc.oc.cox.net) |
| 14:47.05 | [TK]D-Fender | merlin8282, No, we're saying stop using named extens. |
| 14:47.13 | p3nguin | The other method recommended is different from my way, but will yield the same results. |
| 14:47.19 | [TK]D-Fender | merlin8282, and starrt using labels properly |
| 14:47.27 | pdtpatrick1 | Question .. google.com/calendar/dav/username@gmail.com/events no longer works? |
| 14:47.34 | *** join/#asterisk jhoppe (~jhoppe@68-188-9-110.static.stls.mo.charter.com) |
| 14:47.41 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:47.43 | pdtpatrick1 | keeps asking for username and password but does not allow one in.. however i used ical |
| 14:47.46 | pdtpatrick1 | which worked once |
| 14:47.49 | pdtpatrick1 | and then stopped working |
| 14:48.22 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:48.44 | merlin8282 | [TK]D-Fender: you mean like "exten => 1234,n(BUSY)" for example ? |
| 14:48.51 | p3nguin | Are labels case sensitive? |
| 14:48.55 | [TK]D-Fender | merlin8282, yes |
| 14:48.58 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:49.22 | Qwell | AEL would make that so much easier. |
| 14:49.31 | Qwell | switch (${DIALSTATUS}) ... |
| 14:54.24 | pdtpatrick1 | anyone -- google calendar? caldav or ics ? neither is working or does google kick you out if you refresh too often? |
| 14:58.52 | [TK]D-Fender | pdtpatrick1, Umm... you sure you're in the right channel? |
| 14:59.07 | pdtpatrick1 | yup |
| 14:59.13 | pdtpatrick1 | it pulled it before and then it just stopped |
| 14:59.20 | pdtpatrick1 | i can even download the ICS file manually |
| 14:59.34 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 15:00.05 | merlin8282 | http://pastebin.archlinux.fr/434154 <-- did it like this, is this better ? |
| 15:00.22 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 15:00.35 | *** join/#asterisk mocker (~mocker@206.55.118.84) |
| 15:01.48 | p3nguin | merlin8282: The concept looks right. Now decide if you're going to do anything when the status is ANSWER, INVALIDARGS, etc. |
| 15:02.38 | merlin8282 | p3nguin: yes, that's ok. For the moment it's only hangup :) |
| 15:03.02 | merlin8282 | is already familiar with asterisk, but not with all best practices :/ |
| 15:03.05 | p3nguin | Then the others go to voicemail? |
| 15:03.07 | [TK]D-Fender | merlin8282, looks much better |
| 15:03.17 | merlin8282 | p3nguin: right |
| 15:03.19 | p3nguin | Yeah it does. Looks real good. |
| 15:03.46 | [TK]D-Fender | merlin8282, In your specific case you only care about "NOANSWER". This would be better served with a single GotoIf |
| 15:04.02 | [TK]D-Fender | merlin8282, Unless you are expecting to care abou all those others. |
| 15:04.12 | merlin8282 | ah ok, i see |
| 15:04.15 | p3nguin | Let me just paste my exact dial plan using this method. |
| 15:04.44 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:04.48 | hobodave | hey guys |
| 15:05.14 | [TK]D-Fender | or "busy" actually |
| 15:05.43 | Katty | weeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 15:05.49 | Katty | spins in chair |
| 15:05.51 | ollii | hey |
| 15:05.59 | Qwell | leans the chair way back |
| 15:05.59 | hobodave | what format should I be saving my recordings in for Asterisk to be able to read them? I'm using Audacity on OSX. I tried saving first as GSM, but I get a lot of static sound when listening via Asterisk. Now I tried 44.1 kHz WAV (Windows PCM) and I got this error: "Unexpected frequency mismatch 44100 (expecting 8000)" |
| 15:06.03 | Katty | Qwell: :< |
| 15:06.16 | Qwell | Katty: how is that :<?! :( |
| 15:06.18 | p3nguin | http://pastebin.com/5SABNWST |
| 15:06.36 | [TK]D-Fender | merlin8282, http://pastebin.archlinux.fr/434155 |
| 15:06.37 | Katty | Qwell: do not want to go flying |
| 15:06.39 | hobodave | is 8K the best sample rate you can use? Is WAV the best format? |
| 15:06.48 | Katty | p3nguin: are you going to go to any haunted house or halloween stuffs in stl this year? |
| 15:07.06 | p3nguin | I might. I often do. |
| 15:07.12 | [TK]D-Fender | hobodave, WAV is a container, not a specific bitrate, etc |
| 15:07.27 | Katty | p3nguin: a few of us down here are thinking about going too |
| 15:07.28 | hobodave | ok, that makes sense |
| 15:07.34 | Katty | p3nguin: fright fest, or perhaps lemp |
| 15:07.34 | WIMPy | WAV is not even a container |
| 15:07.35 | [TK]D-Fender | hobodave, And *'s mixing core is 8khz 16bit mono. |
| 15:07.40 | hobodave | I'm not very familiar with WAV |
| 15:07.41 | merlin8282 | [TK]D-Fender: yes, that's how I did imagine it, thanks for all your advices :) |
| 15:07.49 | WIMPy | It's a chunk in a RIFF container. |
| 15:07.59 | [TK]D-Fender | hobodave, So best to have in the native formats of whatever kind of calls you'll be processing. |
| 15:08.08 | [TK]D-Fender | hobodave, You don't want * doing any more work than it has to |
| 15:08.53 | [TK]D-Fender | merlin8282, http://pastebin.archlinux.fr/434156 <- I got it backwards |
| 15:10.12 | p3nguin | katty: I heard that the Lemp brewery was open again. Like 10 or 11 years ago they did the "this is the last year, and the steel doors will be welded shut FOREVER" thing, so I had to go that year. |
| 15:10.47 | Katty | p3nguin: i'm hoping the city museum will do something halloweenish |
| 15:11.02 | Katty | p3nguin: might be more kid-friendly too. some of my friends have kids. |
| 15:11.11 | p3nguin | Like me? :) |
| 15:11.14 | Katty | p3nguin: not to mention i don't do horror well...but fright fest would be epic! |
| 15:11.40 | merlin8282 | [TK]D-Fender: yes, hehe. But what now if the callee hangs up, but not the caller ? Does he go to voicemail or is he hung up also ? |
| 15:11.55 | Katty | p3nguin: i'll let you know if any of us head north for halloween |
| 15:12.11 | p3nguin | I can't remember if fright fest was where they had the creature that we had to walk through. |
| 15:12.12 | Katty | p3nguin: i've got a party at my house, but that's a bit of a haul |
| 15:12.20 | Katty | fright fest is the six flags thing |
| 15:12.24 | p3nguin | Oh |
| 15:12.29 | [TK]D-Fender | merlin8282, ... if you answered the call.... then whey would yuo hit voicemail afterwards? |
| 15:12.35 | p3nguin | What the heck am I thinking of, then? |
| 15:12.39 | Katty | no idea |
| 15:13.00 | merlin8282 | [TK]D-Fender: if ${DIALSTATUS} is ANSWER ? |
| 15:13.29 | merlin8282 | the is *not* to reach voicemail, though |
| 15:13.38 | [TK]D-Fender | merlin8282, If you didn't specify a special dial argument, then it will never land on that anyway. And even if you did answer... what more would you want to do? |
| 15:13.59 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:15.37 | p3nguin | Maybe it's Fear Fest. |
| 15:15.54 | merlin8282 | I meant "the aim is *not* to reach voicemail". Anyway. I just try to think of all possibilities, in order to avoid strange reactions from asterisk. |
| 15:16.11 | p3nguin | http://www.fearfesthauntedhouse.com/ |
| 15:17.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:17.29 | Katty | oooh |
| 15:17.29 | [TK]D-Fender | merlin8282, unless you tell Dial otherwise, it will kill the call after a hangup. So no need to worry about that dialstatus for this purpose |
| 15:17.47 | p3nguin | I've been to so many different ones over the past 15 years, I don't remember all the names and locations. |
| 15:17.53 | merlin8282 | [TK]D-Fender: okay. |
| 15:18.04 | Katty | p3nguin: i've never been to a haunted house |
| 15:18.07 | Kobaz | woah it's [TK]D-Fender |
| 15:18.11 | p3nguin | GASP! |
| 15:18.19 | coppice | Katty: nobody has |
| 15:18.31 | Katty | coppice: well yes, despite that smalld etail |
| 15:18.39 | Kobaz | it's like when Christopher Walken randomly shows up in a movie |
| 15:19.00 | Katty | <3 christopher walken |
| 15:19.09 | coppice | he should do Sesame Street |
| 15:19.31 | *** join/#asterisk atheos (~atheos@208.119.68.16) |
| 15:19.54 | coppice | he seems to have become a parody of his younger self |
| 15:20.05 | Kobaz | it's like... holy $(*&#$!! it's Christopher Walken |
| 15:20.18 | Kobaz | we need a holy $(*&#$!! it's [TK]D-Fender t-shirt |
| 15:20.28 | Katty | ohh zombie safari! |
| 15:20.34 | Katty | paintball sounds fun for halloween |
| 15:20.55 | Kobaz | what about an airsoft sniper |
| 15:20.59 | p3nguin | I think that's the same location that had the hayride and stuff that I went to a few years ago. |
| 15:21.08 | *** part/#asterisk charley (boise@epicboise.com) |
| 15:21.26 | p3nguin | Looks like they have changed the stuff a bit, but might be the same place. |
| 15:22.19 | p3nguin | Hill House was pretty okay, but I think it's gone. I took my brother there and some monster chased him with a chainsaw. |
| 15:22.41 | [TK]D-Fender | Kobaz, No, that would lead to bad places... |
| 15:23.01 | Katty | i'm pretty sure if someone chased me with a chainsaw i'd end up breaking their arm |
| 15:23.25 | Kobaz | heh |
| 15:23.26 | [TK]D-Fender | p3nguin, If you haven't seen American Psycho then prepare to ruin your other chainsaw-related imagery. |
| 15:23.45 | p3nguin | That movie was on the other night, but I didn't watch it. |
| 15:23.59 | [TK]D-Fender | you should have... |
| 15:24.11 | Kobaz | http://www.husqvarna.com/us/homeowner/accessories/other-accessories/practical-items/toy-chain-saw/ |
| 15:24.14 | [TK]D-Fender | Go grab it when you get the chance |
| 15:24.40 | coppice | who was the guy who used to juggle chainsaws? |
| 15:24.50 | p3nguin | There are several of them. |
| 15:25.01 | [TK]D-Fender | coppice, Dunno, but he was awesome. Somebody should give him a hand... |
| 15:25.13 | coppice | the one who also juggled margarine |
| 15:25.36 | [TK]D-Fender | coppice, lol |
| 15:25.44 | p3nguin | Because butter was bad for his health? |
| 15:25.45 | Qwell | I watched Kobaz juggle chainsaws at astricon last year. True story. |
| 15:25.57 | Qwell | He was pretty drunk though and probably doesn't remember that. |
| 15:26.28 | coppice | well, a dollop of 3 or 4 litres of margarine, a squashed ball of bread, and a fresh chicken |
| 15:28.41 | p3nguin | That's one hell of a dollop. |
| 15:29.46 | coppice | well, he needed something roughly chicken sized. he called it juggling food groups - meat, carbohydrates, and petrochemicals |
| 15:29.56 | Katty | *hee* |
| 15:31.19 | *** join/#asterisk m_tadeu (~quassel@89-180-77-153.net.novis.pt) |
| 15:38.46 | Kobaz | hehe |
| 15:39.07 | Kobaz | sounds about right |
| 15:40.40 | *** join/#asterisk fecal (~jerware@c-174-54-171-178.hsd1.pa.comcast.net) |
| 15:40.42 | fecal | hi |
| 15:41.01 | [TK]D-Fender | And post-food groups.... |
| 15:41.12 | fecal | what is the role of Asterisk? Why can't one just buy an IP phone and plug it into the local switch? |
| 15:41.24 | Qwell | fecal: What is the "switch" going to talk to? |
| 15:41.30 | ollii | <3 |
| 15:41.32 | fecal | the default gateway. |
| 15:42.02 | ollii | fecal: asterisk is a pbx...an ip phone not |
| 15:42.18 | fecal | If I have an VoIP provider, will I still need a pbx ? |
| 15:42.37 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 15:43.04 | ollii | fecal: http://en.wikipedia.org/wiki/Private_branch_exchange#Private_branch_exchange |
| 15:43.06 | ollii | decide |
| 15:43.07 | [TK]D-Fender | fecal, why do you need a trampoline? |
| 15:43.55 | [TK]D-Fender | fecal, A "VoIP provider" (gneric enough as it is), is not a PBX. They are not substitutes for one another. They are completely different |
| 15:45.17 | *** join/#asterisk epaphus (~Propietar@201.199.62.74) |
| 15:45.43 | epaphus | hello. Asterisk is a Class 5 PBX right ? |
| 15:45.51 | Qwell | epaphus: it can be |
| 15:46.10 | coppice | Class 5 != PBX |
| 15:46.15 | Qwell | (in the same way that a lego isn't a spaceship) |
| 15:46.38 | epaphus | what Class is a trunk ? |
| 15:50.35 | [TK]D-Fender | <epaphus> what Class is a trunk ? <- not a valid question. |
| 15:50.44 | epaphus | :/ |
| 15:50.57 | [TK]D-Fender | "trunk" is too generic a term just for starters. |
| 15:50.58 | epaphus | Where canI read more about Classes ? :P |
| 15:51.07 | [TK]D-Fender | School? :p |
| 15:51.16 | epaphus | haha. |
| 15:51.32 | [TK]D-Fender | epaphus, http://www.google.ca/#sclient=psy-ab&hl=en&biw=1920&bih=1112&source=hp&q=telephony+switch+classes&pbx=1&oq=telephony+switch+classes&aq=f&aqi=q-w1&aql=1&gs_sm=e&gs_upl=2379l7303l0l7541l26l23l1l1l1l1l538l6872l2-6.12.1.1l22l0&bav=on.2,or.r_gc.r_pw.&fp=3647ad0517bd73b8 |
| 15:51.56 | [TK]D-Fender | epaphus, Google shows an amazingly suggestive pile of suggestions on 3 keywords... |
| 15:52.40 | epaphus | amazing :) thanks |
| 15:55.03 | [TK]D-Fender | Class 5 reads as "Class 5 exchanges were those to which subscribers and end-users telephone lines would connect.". |
| 15:55.48 | [TK]D-Fender | Which on some scale can resemble what * does as a PBX toolkit. You can connect phones to it. |
| 15:56.24 | [TK]D-Fender | However * does not implicitly blong in the chain going upward per-se |
| 15:57.28 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:57.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:11.15 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 16:11.39 | fecal | Which protocol is used for end points(phones) on the same side internal side of a pbx ? And which protocol(s) is used from pbx to VoIP provider. |
| 16:12.11 | WIMPy | yes |
| 16:12.19 | *** join/#asterisk mocker (~mocker@206.55.118.84) |
| 16:12.38 | [TK]D-Fender | fecal, there are MANY different protocols. Asterisk speaks many of them |
| 16:13.10 | Katty | chuckf: run, forest, run! |
| 16:13.28 | [TK]D-Fender | fecal, Here are jsut a few VoIP protocols : SIP, MGCP, IAX2, H.323, Skype, and more |
| 16:14.00 | *** join/#asterisk brezular (~brezular@adsl-dyn253.78-98-90.t-com.sk) |
| 16:16.00 | chuckf | Katty: what direction? |
| 16:16.02 | [TK]D-Fender | fecal, * can speak any protocol to any kind of thing on the other side and pass calls between them |
| 16:17.06 | Katty | chuckf: did you get a shower? |
| 16:18.07 | chuckf | I'm going to say I showered this morning with the caveat that i've not been following the channel discussions today very close |
| 16:18.14 | [TK]D-Fender | fecal, I recommend you download the book and get a basic understanding of VoIP & TDM telephony and understanding Asterisk's role in it. |
| 16:18.16 | [TK]D-Fender | ~book |
| 16:18.17 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 16:19.05 | Katty | chuckf: i was just poking at you since nike+ posted you ran for 8 miles |
| 16:19.44 | [TK]D-Fender | leifmadsen, is 3rd Ed released in a freely downloadable PDF format somewhere? |
| 16:19.48 | chuckf | ah, okay. Now I get it |
| 16:19.54 | Katty | :P |
| 16:20.28 | chuckf | Katty: that wasn't me. My wife has somehow gotten her nike+ account syncing through my fb account |
| 16:21.01 | chuckf | Every time I see those posted I tell her to fix it, and you see how well she has done that |
| 16:21.05 | d_preston215 | I'm getting this when trying to make a call: |
| 16:21.08 | d_preston215 | <PROTECTED> |
| 16:21.37 | [TK]D-Fender | d_preston215, http://networking.ringofsaturn.com/Routers/isdncausecodes.php |
| 16:21.38 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 16:21.40 | Katty | chuckf: so then she needs a shower |
| 16:21.44 | Katty | chuckf: you should go help her with that |
| 16:22.03 | chuckf | I'd like to, but I'm a bit far to do that at the moment |
| 16:22.16 | Katty | :< |
| 16:22.20 | WIMPy | d_preston215: {0x1B, "Destination out of order"}, |
| 16:39.43 | *** join/#asterisk bchia (~Adium@nat/digium/x-rvmsytyilzmcrorg) |
| 16:40.52 | leifmadsen | [TK]D-Fender: no, just HTML |
| 16:47.13 | *** join/#asterisk irroot (~irroot@41.51.134.145) |
| 16:47.16 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:47.35 | *** join/#asterisk imox (~imox@p4FC5C7C5.dip0.t-ipconnect.de) |
| 16:50.11 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 16:50.14 | [sr] | gray |
| 16:50.16 | [sr] | hi WIMPy |
| 16:50.32 | WIMPy | hi [sr] |
| 16:53.28 | [sr] | whats up |
| 16:57.08 | citywok | word |
| 16:57.11 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:02.01 | [sr] | im tired |
| 17:02.05 | [sr] | going home |
| 17:02.58 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176002162.dsl.bell.ca) |
| 17:07.31 | WIMPy | ~gtalk |
| 17:07.35 | Naikrovek | leifmadsen: why html this time? is that because of planned changes/corrections? |
| 17:07.36 | *** part/#asterisk imox (~imox@p4FC5C7C5.dip0.t-ipconnect.de) |
| 17:07.47 | WIMPy | ~gvoice |
| 17:07.47 | infobot | Voice control for Gtk/GNOME applications. URL: http://www.cse.ogi.edu/~omega/gnome/gvoice/ |
| 17:07.55 | [TK]D-Fender | Naikrovek, I'd be betting to promote the ebook download sales |
| 17:08.10 | Naikrovek | [TK]D-Fender: that was my next question. |
| 17:08.10 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 17:08.15 | leifmadsen | Naikrovek: yes, and because O'Reilly needed us to not provide the PDF for other reasons (other agreements they have with people like Safari etc) |
| 17:08.25 | Naikrovek | fair enough. |
| 17:08.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:08.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:08.40 | Naikrovek | I did love the PDF though. Browsers can't render text very well, compared to a PDF. |
| 17:08.56 | leifmadsen | you're welcome to support the authors with a purchase of the PDF then :) |
| 17:09.11 | Naikrovek | leifmadsen: I can virtually guarantee it. |
| 17:09.15 | leifmadsen | :) |
| 17:09.30 | leifmadsen | marks another nickel on his chalkboard |
| 17:09.55 | Naikrovek | since when does Safari offer PDFs anyway? They only had weak HTML only layouts last I checked. |
| 17:10.02 | Naikrovek | looked awful |
| 17:10.17 | leifmadsen | Naikrovek: that was just an example, not the definitive list of reasons |
| 17:10.25 | Naikrovek | the example is fine |
| 17:10.31 | Naikrovek | i don't need to see your contract, heh |
| 17:10.34 | Naikrovek | just curious |
| 17:10.38 | leifmadsen | regardless, O'Reilly wanted to sell electronic copies, and we weren't opposed to that |
| 17:12.54 | JonathanRose | p3nguin: Had an opportunity to try out elguero's new patch for Asterisk-18626 yet? |
| 17:14.07 | russellb | leifmadsen: for me it was more about wanting to ensure that people got the newest content, and also promoting the feedback loop with ofps |
| 17:14.39 | leifmadsen | russellb: yep that was definitely an advantage. We can now push the resolved errata to the web right away, which reminds me I should go push that button :) |
| 17:14.43 | leifmadsen | I fixed some errata this morning |
| 17:14.46 | russellb | cool. |
| 17:15.57 | Naikrovek | the instantness of that must be nice |
| 17:16.52 | leifmadsen | when it works ya :) |
| 17:16.54 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 17:23.23 | *** join/#asterisk irroot (~irroot@197.106.50.10) |
| 17:25.24 | leifmadsen | Naikrovek: ya we just make the changes in SVN, then merge to the trunk, and then hit a button the dashboard |
| 17:27.46 | *** join/#asterisk r1ppa (~McBoingBo@mail.hrsg.ca) |
| 17:29.20 | dijib | anybody know how to install res_meetme.so ? |
| 17:29.41 | r1ppa | Been getting some complaints lately about audio quality with softphones, using X-Lite, as a sysadmin I know that it is most likely that they are pushing VOIP traffic over large distances and on VPN...but I need to get some conclusive data, can someone help troubleshoot Asterisk quality problems, there must be some tools or methods to use that make it more obvious |
| 17:30.51 | r1ppa | also, I dont know if it is simply because X-Lite is free garbage, but trying the audio test, it is telling me audio quality poor/limited data sent/received/possible codec mismatch... |
| 17:34.22 | [TK]D-Fender | MeetMe is a "res" now, and not an "app"? |
| 17:34.35 | [TK]D-Fender | O>o |
| 17:34.45 | dijib | ok but where do i get the .so |
| 17:35.03 | [TK]D-Fender | dijib, Should be right in there with all the other apps |
| 17:35.04 | dijib | i installed from yum and its not in /usr/lib/asterisk/modules |
| 17:35.37 | [TK]D-Fender | dijib, Perhaps you were missing having installed DAHDi first as that is a pre-requisite for that specific app... |
| 17:35.56 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 17:36.02 | dijib | ahhh that must be it then |
| 17:36.18 | dijib | when i do that i get a kmod dependency error i cant seem to resolve |
| 17:36.45 | [TK]D-Fender | Keep working on it.. |
| 17:37.08 | dijib | are we talking dahdi-linux or asterisk18-dahdi |
| 17:37.09 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 17:37.11 | dijib | or both? |
| 17:37.15 | [TK]D-Fender | both |
| 17:40.01 | r1ppa | Is it a bad idea VPN + VOIP with a softphone? |
| 17:40.18 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 17:41.22 | [TK]D-Fender | r1ppa, Doesn't matter as long as packets arrive when they should |
| 17:42.17 | r1ppa | how do I determine a problem if there are any? |
| 17:42.34 | [TK]D-Fender | r1ppa, You'll hear it |
| 17:42.52 | [TK]D-Fender | r1ppa, If you can't hear the difference, then there isn't enough of one to make one :) |
| 17:43.26 | r1ppa | I just dont know where to start, I am fairly certain the real issue is lack of bandwidth, but I want some evidence, and to understand how to troubleshoot potential audio call problems |
| 17:44.00 | [TK]D-Fender | r1ppa, You seem to have "actual "problems more than just "potential ones". |
| 17:44.13 | pervertedjustice | hi |
| 17:44.13 | [TK]D-Fender | r1ppa, And your first test would be to pass that call off outside of the VPN |
| 17:44.14 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:44.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:44.17 | r1ppa | for instance, there was a conference several days ago where audio was terrible at times and callers were getting dropped, I had no reason for them, Asterisk was happy |
| 17:44.37 | [TK]D-Fender | if using the same link you get the same result then that removes VPN as a layer at fault |
| 17:44.53 | [TK]D-Fender | Bandwidth is certainly important. |
| 17:44.59 | r1ppa | but without VPN there is NO way to the VOIP server |
| 17:45.02 | [TK]D-Fender | and being choked will kill calls, etc |
| 17:45.14 | r1ppa | we didnt poke a hole to the outside world from the Asterisk server |
| 17:45.22 | [TK]D-Fender | r1ppa, then I supposed you'd better set up an alternative if you want to rule it out. |
| 17:45.33 | r1ppa | true enough |
| 17:45.51 | [TK]D-Fender | Didn't poke a hole? Well go do it. |
| 17:45.54 | r1ppa | I guess I am doing what users do to me, not getting to the point lol |
| 17:45.59 | [TK]D-Fender | It what you've got to do.. |
| 17:46.20 | [TK]D-Fender | r1ppa, The funny thing is you alrady know most of all these things you should be doing... |
| 17:46.35 | r1ppa | I really just want best practices on what to monitor to catch audio problems |
| 17:47.10 | [TK]D-Fender | r1ppa, best practice = strip out all the variables one by one examining closely as you do to see what the weights are to each of the factors in play |
| 17:47.21 | [TK]D-Fender | r1ppa, Basic scientific process. |
| 17:48.24 | *** join/#asterisk irroot (~irroot@41.52.186.3) |
| 17:48.26 | r1ppa | "why did calls get dropped"....I dont know, bandwidth was fine, load was fine, ping to VOIP provider and ISP was fine...Asterisk log was only complaining about the channels getting full, as we only have like 8-10 channels |
| 17:49.09 | *** join/#asterisk bmg505 (~leon@196-209-44-105.dynamic.isadsl.co.za) |
| 17:49.41 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:50.10 | citywok | r1ppa: i find taking a packet capture on the asterisk box and then listening to the RTP streams using wireshark is a good place ot start for identifying audio problems. |
| 17:50.34 | citywok | it helps me identify which leg of the call is the problem at the source, and since my saterisk box is in a datacenter and all my phones are in other offices it makes quite a difference. |
| 17:50.50 | Katty | i sneezed. |
| 17:50.56 | citywok | bless you |
| 17:50.58 | Katty | and it hurt. |
| 17:51.02 | Katty | ty |
| 17:51.10 | Katty | don't you hate it when sneezes hurt |
| 17:51.20 | Katty | fine one second, the next... OHGODMYABS |
| 17:51.25 | citywok | i've only had them hurt when i did an ab day |
| 17:51.46 | *** join/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 17:51.48 | citywok | and then it's holy crap i hurt for a minute. same for coughing, or laughing after a brutal ab day |
| 17:52.11 | irroot | Katty reminds me i need to clean my screen .... hate it wih it had windscreen wipers |
| 17:52.16 | Katty | every day is ab day! |
| 17:52.18 | r1ppa | any preference of softphone for Asterisk guys? Using X-lite, but more recently EyeBeam (because of the G729 codec of course) |
| 17:52.50 | citywok | Katty: my abs can't handle that. today was the first day at the gym in a month (2 week vacation of backpacking & wakeboarding was good enough). chest day, and it was pathetic. |
| 17:53.00 | citywok | ~softphones |
| 17:53.05 | Katty | aww. |
| 17:53.07 | citywok | ahh, there isn't one |
| 17:53.21 | Katty | i don't have a Chest Day |
| 17:53.26 | Katty | unless pushups count |
| 17:53.30 | citywok | i'm afraid for back & shoulders day. deadlifts are going to be awful. |
| 17:53.36 | citywok | pushups do count, but pushups don't make a day : |
| 17:53.40 | citywok | :P |
| 17:53.43 | Katty | but you're a boy |
| 17:53.47 | Katty | you have boy parts. |
| 17:53.52 | Katty | they work differently |
| 17:53.55 | citywok | how do you know that?!?!?!? |
| 17:54.07 | Katty | cause you're talking about chest, back, and shoulders day |
| 17:54.16 | Katty | these are not days women have. |
| 17:54.37 | citywok | hahaha, okay, for a second i thought i was going to have to look for a hidden camera. *phew* |
| 17:54.47 | citywok | there's also arms day, and legs day! |
| 17:54.48 | tzanger | ... abs? |
| 17:54.56 | tzanger | I've hurt my ribs sneezing but never my ribs |
| 17:54.58 | citywok | tzanger: those things behind the beer belly |
| 17:54.58 | tzanger | haha |
| 17:55.05 | tzanger | I've hurt my ribs sneezing but never my abs |
| 17:55.23 | leifmadsen | what's an abs? |
| 17:55.26 | citywok | really? after a really good ab day, think when it's hard to sit up when you're in bed, it can be brutal |
| 17:55.40 | Katty | leifmadsen: it's that thing you do when someone acts like they're going to punch you in the gut |
| 17:55.48 | citywok | haha |
| 17:55.54 | [TK]D-Fender | <Katty> i don't have a Chest Day <- you're female. EVERY day is "chest day" ;) |
| 17:55.56 | leifmadsen | Katty: I work from home alone -- that doesn't happen to me |
| 17:56.11 | tzanger | chesty laroux? |
| 17:56.24 | Katty | leifmadsen: it's that thing you do when you're sitting in a chair, and a 20lb cat jumps up and threatens to knock you balance out of wack |
| 17:57.02 | Katty | [TK]D-Fender: dont' be redonkulious. |
| 17:57.13 | leifmadsen | Katty: I dont' have any cats :) |
| 17:57.26 | leifmadsen | tzanger: Hooty McBoob? |
| 17:57.26 | citywok | leifmadsen: it's what the beer belly insulates :P |
| 17:57.33 | leifmadsen | citywok: oh got it |
| 17:58.04 | Katty | i don't think leif has one of those |
| 17:58.21 | Katty | checks |
| 17:58.25 | citywok | everybody has a beer belly. it's a question of full keg, quarter barrel, pony keg |
| 17:58.33 | citywok | i on the other hand only have a six pack ;) |
| 17:58.43 | leifmadsen | citywok: I will beat you up at AstriCon regardless |
| 17:58.52 | citywok | nooooo |
| 17:58.52 | leifmadsen | who said that?! |
| 17:59.05 | citywok | but i dun wanna get beat up :( |
| 17:59.41 | _Corey_ | "Fight Club" at Astricon? Nice |
| 18:00.09 | Katty | we don't talk about fight club |
| 18:00.15 | citywok | but how will you and russell have a rematch this year if russell isn't there?!? |
| 18:00.27 | Katty | i'll take his place |
| 18:00.33 | Katty | i'd like to see leif try to beat me up |
| 18:00.38 | Qwell | me too |
| 18:00.41 | Katty | that'd be a hilarious sight |
| 18:00.43 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 18:00.46 | Qwell | leifmadsen'd get his ass kicked. |
| 18:00.51 | citywok | hahaha |
| 18:00.58 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:01.33 | Katty | i havin knittin needles and i know how to use em |
| 18:02.05 | Katty | that sweater has NO CHANCE |
| 18:02.22 | citywok | knit that thing to pieces! |
| 18:02.31 | Katty | *hee* |
| 18:03.39 | Katty | heaven help me if i ever get into a real fight. |
| 18:03.58 | Katty | my scream might break all the windows. |
| 18:04.00 | citywok | yea... i'd like to avoid it if at all possible. |
| 18:04.34 | Katty | tis a good thing to avoid. |
| 18:04.40 | Katty | much better things to be doing. like drinking. |
| 18:04.46 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 18:04.50 | Katty | and pestering the crap out of Qwell |
| 18:04.55 | Qwell | huh what? |
| 18:05.03 | citywok | Yes, drinking. and not working. and maybe working out sometimes so if i do get in a fight i don't get murdered. |
| 18:05.04 | Katty | exactly, see! |
| 18:05.07 | Qwell | is "pestering" code for something? |
| 18:05.13 | citywok | yes. poking. |
| 18:05.29 | Katty | i've been known to poke people. |
| 18:06.07 | Qwell | tw...h...s? |
| 18:06.14 | citywok | :D |
| 18:06.42 | Katty | i'd like to buy a vowel. |
| 18:06.46 | *** join/#asterisk jkroon (~jkroon@dsl-241-237-66.telkomadsl.co.za) |
| 18:06.54 | citywok | fu isn't a vowel |
| 18:07.30 | Katty | depends on how you say it. |
| 18:11.34 | dijib | if i have installed asterisk through a package manager, and then reinstall through a sroucefile would it just overwrite the existing? |
| 18:11.59 | dijib | i cant get these dependencies resolved |
| 18:12.20 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:12.47 | Katty | somewhere around here i have a list of dependencies i apt-get before the tarballs |
| 18:12.56 | Katty | would you like me to pastebin it? |
| 18:13.05 | dijib | would love you too |
| 18:13.10 | Katty | mkay |
| 18:13.13 | dijib | thanks |
| 18:13.48 | irroot | Katty glibc on that list :P |
| 18:14.59 | Katty | dijib: http://pastebin.com/F8gRPxbX |
| 18:15.13 | Katty | dijib: not all of them are dependencies.. it's just Katty's-List-O-Stuff |
| 18:15.55 | dijib | thanks katty |
| 18:16.14 | Katty | mhmm |
| 18:16.46 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 18:20.49 | citywok | Katty: your list looks similar to mine although i pipe the current uname in to the kernel-headers so i don't have to worry about it. |
| 18:21.10 | citywok | brb |
| 18:22.05 | *** join/#asterisk irroot (~irroot@197.107.38.171) |
| 18:22.56 | dijib | Katty, a lot on that list i dont need |
| 18:22.58 | dijib | but thanks |
| 18:23.32 | *** join/#asterisk Godfather_ (~estanteri@58.Red-88-5-38.dynamicIP.rima-tde.net) |
| 18:23.46 | Katty | no problemo |
| 18:25.41 | irroot | nice job Kattu |
| 18:25.53 | irroot | s/Kattu/Katty/ |
| 18:29.31 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 18:32.08 | p3nguin | dijib: No, it will not overwrite. |
| 18:33.08 | p3nguin | dijib: Source install should go to a different prefix than the location of package install. |
| 18:34.02 | p3nguin | Pipe the current uname in to the kernel-headers? Huh? |
| 18:34.23 | p3nguin | uname|kernel-headers |
| 18:34.30 | p3nguin | Does not compute. |
| 18:35.21 | p3nguin | You know what would be neat was if there were some way for dependencies to be solved automatically when you install a package. |
| 18:35.22 | *** join/#asterisk resno (~resno@unaffiliated/resno) |
| 18:35.47 | resno | i want to setup asterisk, but the conf files are a bit confusing for me. any suggestions on a web frontend that works well? |
| 18:35.56 | p3nguin | ~freepbx |
| 18:35.56 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 18:36.18 | p3nguin | But you're going to give up the power for the convenience. |
| 18:36.42 | resno | p3nguin: let me throw in one more gotcha. i trying to virualize it |
| 18:36.43 | [TK]D-Fender | GUI's are also a poor basis to learn * from after. |
| 18:36.47 | resno | i nkow timing issues |
| 18:36.54 | resno | i know* timing issues |
| 18:37.11 | citywok | p3nguin: dev linux-headers-`uname -r` |
| 18:37.13 | [TK]D-Fender | resno, don't worry so much about those. |
| 18:37.17 | p3nguin | That's not a pipe. |
| 18:37.24 | citywok | p3nguin: yea, my bad :P |
| 18:37.30 | p3nguin | Forgiven. |
| 18:37.31 | citywok | so sue me |
| 18:37.32 | *** join/#asterisk irroot (~irroot@197.108.10.147) |
| 18:37.35 | citywok | okay, thanks! |
| 18:37.37 | p3nguin | I SAID FORGIVEN. |
| 18:37.38 | [TK]D-Fender | resno, Go download AsteriskNOW and pump up an image using FreePBX as your GUI and you'll haev something to start with. |
| 18:37.43 | citywok | yea, i hit enter at the same time you did :P |
| 18:37.49 | p3nguin | Forgiven again. |
| 18:37.52 | citywok | ty |
| 18:37.57 | resno | [TK]D-Fender: can it be installed in centos? |
| 18:38.08 | p3nguin | No. It IS CentOS. |
| 18:38.12 | citywok | resno: asterisknow is centos |
| 18:38.13 | resno | as in, insdie of of centos |
| 18:38.13 | [TK]D-Fender | resno, Yes if you want to do it by hand : www.freepbx.org |
| 18:38.38 | p3nguin | AsteriskNOW is a CentOS-based full distro containing asterisk and your choice of Digium GUI, FreePBX, or no GUI. |
| 18:38.38 | [TK]D-Fender | resno, You'll have to install Asterisk, FreePBX, prep up MySQL, etc |
| 18:38.44 | [TK]D-Fender | resno, Apache, etc |
| 18:39.20 | resno | well, do youll know of a container that hasasterisknow? |
| 18:39.28 | resno | wow, that ran togeher |
| 18:39.33 | [TK]D-Fender | resno, Also normally you run apache as user "asterisk" which means don't do this on a machine hosting web pages, etc |
| 18:40.06 | resno | right, which is why im trying to viruzliate it |
| 18:40.10 | p3nguin | Just grab the AsteriskNOW iso. |
| 18:40.12 | resno | virutalize it |
| 18:40.14 | p3nguin | Then install it. |
| 18:40.33 | resno | you're sending me down a street i dont want to go down |
| 18:40.40 | p3nguin | It'll install in a virtual machine the same as any other OS. |
| 18:40.50 | p3nguin | It's a 15-minute procedure. |
| 18:40.54 | resno | i am using containers |
| 18:41.04 | p3nguin | You've spent almost that much time asking questions about how to not install it. |
| 18:41.09 | resno | my machine wont accept isos |
| 18:41.25 | [TK]D-Fender | resno, Go to their page. Follow the instructions. |
| 18:41.29 | p3nguin | There's a tool out there somewhere to make your container from a CD image. |
| 18:41.53 | resno | returns to google |
| 18:42.00 | resno | heh, that was fun |
| 18:42.19 | [TK]D-Fender | resno, www.freepbx.org |
| 18:42.23 | [TK]D-Fender | resno, not "google" |
| 18:43.34 | resno | k thanks |
| 18:43.42 | resno | sips coffee |
| 18:45.06 | dijib | Requires: kernel-i686 = 2.6.18-274.3.1.el5 |
| 18:45.07 | dijib | <PROTECTED> |
| 18:45.19 | dijib | harry carry |
| 18:46.38 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053181.ics.uci.edu) |
| 18:50.13 | Qwell | dijib: What did you try to install? |
| 18:50.32 | dijib | dahdi-linux and asterisk18-dahdi |
| 18:50.43 | [TK]D-Fender | Qwell, he wants to install MeetME but got some dep issues for DAHDI |
| 18:50.46 | Qwell | and it's failing, or what? |
| 18:50.53 | [TK]D-Fender | Qwell, which is what he's trying to iron out |
| 18:51.24 | dijib | it says that there is installed kernels, that are newer |
| 18:51.25 | p3nguin | I would have expected "yum -y install dahdi-linux" to take care of the problem. |
| 18:51.27 | Qwell | I haven't seen anything useful yet. |
| 18:51.38 | *** join/#asterisk brezular (~brezular@adsl-dyn155.78-98-114.t-com.sk) |
| 18:51.44 | Qwell | dijib: pastebin the output |
| 18:52.37 | dijib | http://pastebin.com/3v84pgUv |
| 18:53.35 | Qwell | CentOS 6? |
| 18:53.38 | Qwell | walks away |
| 18:53.41 | dijib | yessur. |
| 18:53.50 | citywok | it wants an older kernel than is available |
| 18:53.50 | dijib | using the centos5 repo for asterisk |
| 18:53.57 | dijib | yep |
| 18:53.59 | dijib | thats what i see |
| 18:54.00 | citywok | it wants 2.6.18 whereas all it has avail is 2.6.32 |
| 18:54.09 | citywok | then i guess you'll be compiling it from source ;) |
| 18:54.10 | dijib | so i need to install old kernel ? |
| 18:54.27 | dijib | but if i do that it didnt run when set with chkconfig asterisk on |
| 18:54.28 | Qwell | You need to not use CentOS 6 if you want to use "the centos5 repo for asterisk" |
| 18:55.11 | citywok | lol, but qwell, why!??!? |
| 18:55.13 | dijib | p3nguin, what do you think? |
| 18:55.14 | [TK]D-Fender | dijib, You really, really shouldn't be trying to insist on pouring diesel fuel into a gasoline car.... |
| 18:55.28 | dijib | yeh but it should work! |
| 18:55.35 | citywok | no, no it really shouldn't... |
| 18:55.39 | *** mode/#asterisk [-o Qwell] by Qwell |
| 18:55.42 | citywok | 6 != 5 |
| 18:55.44 | Qwell | for your safety. |
| 18:55.59 | dijib | and i have no choice... it was the only thing that would run on this old p4 laptop. Centos4&5 dont run, asterisk now doesnt. so on. |
| 18:56.02 | dijib | centos6 does |
| 18:56.38 | Qwell | then you can't use packages.asterisk.org |
| 18:56.39 | dijib | so if i build from source, how do i have it overwrite the copy install from package manager? |
| 18:56.55 | dijib | then i need to find an el6 repo. |
| 18:56.57 | dijib | i suppose |
| 18:57.12 | [TK]D-Fender | dijib, remove them via RPM, trash the modules folder, then steamroll your source right over it |
| 18:57.58 | citywok | didnot: remove the old install. |
| 18:58.08 | citywok | s/didnot/dijib/ |
| 18:58.31 | citywok | russellb: thanks for the present! |
| 18:58.37 | dijib | last time i did the source install it would not boot as much as i kicked it at startup |
| 18:58.55 | dijib | or im just a useless nuub |
| 18:59.04 | dijib | which is more likely, ask p3nguin |
| 19:01.36 | [TK]D-Fender | "boot" |
| 19:02.06 | [TK]D-Fender | I would love to know how Asterisk installed by source would kill your abiilty to boto the OS you had in place and installed it on... |
| 19:02.49 | citywok | [TK]D-Fender: yea... that sounds like bad terminology or a "useless nuub" |
| 19:03.34 | [TK]D-Fender | I don't know. It was his box in a state, in a point in time. I suppose noone will ever really know. |
| 19:04.33 | [TK]D-Fender | dijib, anyway this appears to be the option you've got. You've layed out what your strict requirements were and have ben given the options that should work with it. Go give it a whirl and we'll see what comes out of it. |
| 19:05.34 | dijib | im eraseing now |
| 19:05.42 | dijib | then kill modules |
| 19:05.45 | p3nguin | I've given up on trying to help him. He always does whatever he wants regardless of the best efforts of others to steer him in the right direction. |
| 19:06.04 | dijib | i beg to differ entirely p3nguin |
| 19:06.13 | p3nguin | You would. |
| 19:06.22 | dijib | nope. incorrect sir |
| 19:06.36 | dijib | im presently on a path to install from source |
| 19:06.45 | citywok | lol. p3nguin at some point you have to cut your losses and run. like the guy last night that couldn't get his groupings to work even after tk wrote the code for him. |
| 19:06.56 | p3nguin | Yep, exactly like that. |
| 19:07.42 | [TK]D-Fender | citywok, He came back with it having failed? |
| 19:08.08 | citywok | [TK]D-Fender: he never tried it b/c he couldn't figure out what config it went in. apparently the wiki article telling you, and just looking at the two config files was too hard. |
| 19:08.25 | citywok | that and he used zaptel/dahdi so interchangeably i really have no idea which one he was using |
| 19:08.37 | citywok | i gave up and went to the bar |
| 19:08.52 | [TK]D-Fender | I told him to kill off the zapata and where to put them.... oh well. |
| 19:09.02 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 19:09.26 | citywok | yea he wasn't interested in doing it. i offered to consult and charge him $125/hr to do it for him. |
| 19:09.55 | citywok | my highlight from yesterday was the trixbox is asterisk and my boss wants trixbox b/c it's open source and peole tell him asterisk is good. |
| 19:13.04 | dijib | p3nguin, i am greatful for all the help you have given. without it i wouldnt have a basic understanding of asterisk that i do |
| 19:13.10 | dijib | thank you. |
| 19:13.37 | citywok | ~book |
| 19:13.37 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 19:14.03 | dijib | i saw that one. |
| 19:14.11 | citywok | i just got my copy today :P |
| 19:14.24 | citywok | ty russell heh |
| 19:14.38 | dijib | i need to learn linux i think before i hardcore digin to asterisk. |
| 19:15.01 | dijib | yum erase asterisk18 asterisk18-addons asterisk18-addons-bluetooth asterisk18-addons-core asterisk18-addons-mysql asterisk18-addons-ooh323 asterisk18-alsa asterisk18-codec_g729a asterisk18-codec_siren14 asterisk18-codec_siren7 asterisk18-configs asterisk18-core asterisk18-curl asterisk18-dahdi asterisk18-devel asterisk18-doc asterisk18-misdn asterisk18-odbc asterisk18-ogg asterisk18-pgsql asterisk18-res_fax_digium asterisk18-r |
| 19:15.01 | dijib | esample asterisk18-skypeforasterisk asterisk18-snmp asterisk18-sqlite3 asterisk18-tds asterisk18-voicemail asterisk18-voicemail-imapstorage asterisk18-voicemail-odbcstorage |
| 19:15.02 | JonathanRose | p3nguin: ping |
| 19:15.03 | dijib | oops |
| 19:15.05 | dijib | sry |
| 19:19.47 | pabelanger | Qwell: haxored? |
| 19:20.29 | dijib | k building |
| 19:20.45 | dijib | garsh darn need a confrence line for thursday. |
| 19:20.52 | dijib | 20min p3nguin |
| 19:21.12 | citywok | dijib: use confbridge instead of meetme which doesn't need dahdi |
| 19:21.34 | dijib | which has more options ? |
| 19:21.42 | dijib | ive already removed all asterisk packages. |
| 19:21.44 | dijib | dont stop me now |
| 19:22.13 | dijib | can you manage attendies the same as in meetme? |
| 19:22.42 | citywok | meetme has more options |
| 19:22.49 | malcolmd | https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 19:23.19 | malcolmd | well...in Asterisk 1.8, confbridge doesn't have a lot of features, that's true |
| 19:24.04 | dijib | im not going to run asterisk 10, too bleeding edge |
| 19:24.29 | citywok | hmm, dang i dont think i ever finished a couple of the commits for confbridge that i was porting from meetme, forgot about them on reviewboard. lol. |
| 19:25.24 | malcolmd | citywok: d'oh :( |
| 19:25.51 | citywok | i think i had written code to do the announce join/leave or at least started it |
| 19:26.22 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 19:26.25 | citywok | that was in december or january though, so i don't really remember. but i got the feature i needed in meetme. that should make 10 |
| 19:30.21 | Katty | hai |
| 19:31.11 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 19:31.16 | devil_evoxxx | hi all |
| 19:31.58 | citywok | hi |
| 19:32.16 | *** join/#asterisk zamba (marius@flage.org) |
| 19:32.45 | zamba | when you have several different devices, smart phones, wired phones and soft phones.. how do you handle it in sip.conf? |
| 19:32.49 | zamba | one entry per defice? |
| 19:32.51 | zamba | device* |
| 19:32.56 | zamba | and then parallell ringing? |
| 19:32.59 | citywok | always one entry per device |
| 19:33.09 | citywok | simultaneous ring you handle in your dial plan (extensions.conf) |
| 19:33.29 | zamba | so you do it that way? |
| 19:33.36 | zamba | simultaneous ringing? |
| 19:33.36 | citywok | yes |
| 19:33.45 | WIMPy | It's the only way |
| 19:33.46 | zamba | ok, cool |
| 19:33.52 | citywok | dial(sip/1593&sip/1550&sip/12535551212@carrier) |
| 19:33.55 | WIMPy | in Asterisk |
| 19:33.56 | zamba | got it |
| 19:34.04 | zamba | @carrier? what does that mean? |
| 19:34.17 | zamba | never seen that before |
| 19:34.27 | [TK]D-Fender | zamba, was a sample of dialing a PSTN number via an ITSP |
| 19:34.29 | citywok | if you want to simulatenous ring your cellphone for example |
| 19:34.37 | zamba | ITSP? |
| 19:34.40 | [TK]D-Fender | ~itsp |
| 19:34.41 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:34.42 | citywok | SIP provider |
| 19:34.54 | zamba | ah, ok |
| 19:34.57 | [TK]D-Fender | citywok, Aim agnostic :) |
| 19:35.00 | citywok | well, SIP is so misused (including by me) |
| 19:35.33 | zamba | that's basically the same as SIP/carrier/number? |
| 19:35.45 | zamba | SIP/number@carrier == SIP/carrier/number? |
| 19:35.52 | citywok | yep |
| 19:35.55 | zamba | ok, cool |
| 19:36.19 | zamba | have you guys used gsm gateways with asterisk |
| 19:36.20 | zamba | ? |
| 19:36.24 | [TK]D-Fender | zamba, Your formatting is the more solid one (SIP/peer/numbertodial) |
| 19:36.47 | citywok | yea, i don't know why i do it the way i do, old habits die hard. |
| 19:37.01 | citywok | maybe b/c i think of local/exten@context |
| 19:37.18 | zamba | http://www.voipon.co.uk/dinstar-dwg2001-dwg20001g-p-3738.html |
| 19:37.21 | zamba | i'm looking at that |
| 19:37.42 | irroot | pabelanger doing some awesome work on configs thx |
| 19:38.03 | [TK]D-Fender | zamba, google up chan_mobile <---- |
| 19:38.18 | citywok | cool, i didn't know they had such a little device. fortunately i've never needed it :P |
| 19:38.22 | [TK]D-Fender | zamba, With a BlueTooth adapter you might be able to use an el-cheapo cellphone |
| 19:38.27 | zamba | [TK]D-Fender: that's a better solution? |
| 19:38.30 | zamba | [TK]D-Fender: ah |
| 19:38.50 | zamba | at least cheaper :) |
| 19:38.53 | [TK]D-Fender | zamba, quite viable, considerably less costly |
| 19:38.58 | WIMPy | Or you use an el cheapo phone via USB cable or an USB data stick. |
| 19:39.12 | citywok | with zamba's thing you could just buy disposable SIM's and have fun heh |
| 19:39.28 | zamba | yeah, using cables for connecting sounds more stable |
| 19:39.47 | zamba | citywok: hehe, yeah.. and you treat it as just another sip peer, as far as i've understood.. so it's much easier |
| 19:40.53 | zamba | [TK]D-Fender: well, i'm looking for something that's production ready and that works totally out of the box without needing compilation |
| 19:42.02 | devil_evoxxx | hi irroot !! |
| 19:42.29 | [TK]D-Fender | zamba, Well a device like that sould do it.. |
| 19:42.32 | irroot | hi there devil_evoxxx |
| 19:42.32 | [TK]D-Fender | should* |
| 19:43.06 | devil_evoxxx | solved the problem with quescom! |
| 19:43.07 | [TK]D-Fender | zamba, Not terribly over-priced all things considered |
| 19:43.11 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 19:43.18 | zamba | [TK]D-Fender: chan_mobile is not an official part of asterisk, so you need to build that in manually, right? |
| 19:43.18 | devil_evoxxx | ...i've replaced it with a beautiful pri card |
| 19:43.41 | [TK]D-Fender | zamba, It is last I checked... |
| 19:44.00 | irroot | devil_evoxxx pri card will give better service and monitoring |
| 19:44.11 | irroot | glad its working for you |
| 19:44.17 | devil_evoxxx | yea..and better debug.. |
| 19:44.32 | devil_evoxxx | when something not work i can see why |
| 19:44.34 | devil_evoxxx | with..quescom |
| 19:44.45 | irroot | you can guess :P |
| 19:44.47 | devil_evoxxx | is like playing with fire.. |
| 19:45.08 | zamba | [TK]D-Fender: not in the ubuntu package of asterisk, at least |
| 19:45.11 | devil_evoxxx | so i've got this when i reload asterisk..(your svn trunk) |
| 19:45.14 | devil_evoxxx | [Sep 27 21:43:51] ERROR[2667]: chan_sip.c:29039 peer_iphash_cb: Empty address |
| 19:45.15 | devil_evoxxx | [Sep 27 21:43:51] ERROR[2667]: netsock2.c:440 ast_sockaddr_hash: Unknown address family '0'. |
| 19:45.39 | [TK]D-Fender | zamba, oh don't even start on those.... |
| 19:46.15 | zamba | [TK]D-Fender: hehe |
| 19:46.38 | zamba | [TK]D-Fender: what's the best distro to be running asterisk then? |
| 19:46.47 | zamba | i'm not too comfortable with building asterisk from source |
| 19:46.49 | citywok | zamba: the one you are familiar with and like -- i personally use debian |
| 19:47.17 | zamba | citywok: do you use debian's apt source for asterisk or do you build it yourself? |
| 19:47.28 | citywok | i compile it, it's very easy |
| 19:47.42 | [TK]D-Fender | zamba, Its not just a question of distro. Ubuntu repacges things themselves. Often sub-standard |
| 19:47.57 | zamba | citywok: well, i was a slackware user in my early days.. there's a reason i've turned from that :) |
| 19:48.26 | zamba | citywok: couldn't trust myself to keep everything patched.. and sources and binaries were laying around all over the place |
| 19:48.29 | [TK]D-Fender | zamba, Same here. I moved to CentOS for servers. I'm happy with Ubuntu for desktops however |
| 19:48.32 | zamba | just a bastard system |
| 19:48.32 | citywok | i use debian only b/c i've been using it ever since i quit using slack / gentoo |
| 19:48.58 | zamba | [TK]D-Fender: you build asterisk from source? |
| 19:49.05 | [TK]D-Fender | zamba, yes |
| 19:49.14 | citywok | and i've found bugs in different versions of asterisk that prevented me from using them, so now i stick to the exact version i've found works best for my needs. |
| 19:49.31 | zamba | [TK]D-Fender: and how do you keep yourself patched? |
| 19:49.34 | citywok | i used packages a long time ago and had a couple issues with the zaptel packages at the time that were unresolved until i installed from source |
| 19:49.37 | [TK]D-Fender | zamba, recompiling. |
| 19:49.52 | citywok | zamba: you compile the new version. but really i just lvae well enough alone. if * is playing nice i avoid upgrading it |
| 19:49.53 | [TK]D-Fender | zamba, 5 mins when the time comes. |
| 19:49.54 | zamba | [TK]D-Fender: yeah, but i mean, you have to watch for bugs and patch manually.. |
| 19:50.30 | [TK]D-Fender | citywok, And then watch a port vulnerability open up and your sever ends up with more holes in it than a #9 sponge :p |
| 19:50.44 | zamba | 1.6.2.5-0ubuntu1.4 |
| 19:50.52 | zamba | that's the one i'm running |
| 19:50.54 | [TK]D-Fender | zamba, OMG... that sounds like responsibility! |
| 19:50.57 | citywok | yea, gotta keep that shit locked down. |
| 19:50.59 | [TK]D-Fender | runs in circles |
| 19:51.25 | [TK]D-Fender | issues a royalty check to file |
| 19:51.33 | zamba | citywok: do you have the sip port open? 5060? |
| 19:51.56 | citywok | unfortunately, yes |
| 19:52.01 | zamba | same here |
| 19:52.08 | citywok | i use fail2ban to watch for auth failures / brute for attacks |
| 19:52.13 | zamba | yup, same here |
| 19:52.22 | citywok | if i wasn't lazy i would use the phone web browser and the polling interval to lock it down |
| 19:52.36 | citywok | have the phone grab a web page, auth that, and then add a firewall rule for the requesting IP |
| 19:53.00 | citywok | alas, i have only thought about doing it, i've never actually implemented it |
| 19:53.01 | zamba | yeah, kind of like knocking |
| 19:53.08 | Katty | hellllllllllllllloooooo nurse. |
| 19:53.12 | zamba | knockd and its corresponding clients |
| 19:53.40 | zamba | what version of asterisk are you running? and what kind of features am i missing out on? |
| 19:53.46 | zamba | which version do you recommend i build from source? |
| 19:54.09 | [TK]D-Fender | zamba, latest 1.8 as per the topic |
| 19:54.43 | zamba | [TK]D-Fender: ok |
| 19:54.56 | zamba | how can i get rid of the "doing dnsmgr_lookup for" messages in the console and in the logs? |
| 19:55.14 | citywok | zamba: i use 1.6.2.11, b/c the times i attempted to upgrdae after that something broke, so i've decided to leave well enough alone |
| 19:55.15 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:55.25 | zamba | citywok: hehe, wise move :) |
| 19:55.34 | citywok | for the installations i've done for small businesses i've been installing 1.8.whatever was current |
| 19:55.40 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 19:56.39 | Katty | has carrot cake |
| 19:57.30 | leifmadsen | p3nguin: ping |
| 19:57.30 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:57.30 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:57.46 | leifmadsen | p3nguin: sounds like there is something ready to be tested on https://issues.asterisk.org/jira/browse/ASTERISK-18626 |
| 20:01.29 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 20:04.24 | Katty | http://a7.sphotos.ak.fbcdn.net/hphotos-ak-ash4/s720x720/301085_903680086077_37617946_39923072_1198111433_n.jpg <- the cake is not a lie! |
| 20:04.27 | Katty | nomnom |
| 20:04.49 | miztic | what kinda cake is that |
| 20:04.57 | Katty | carrot cake |
| 20:05.05 | _Corey_ | hmm, i want cake |
| 20:05.09 | miztic | i knew it!!! vegetables do not belong in cake! |
| 20:05.16 | miztic | that's just wrong |
| 20:05.41 | Katty | pff |
| 20:05.57 | Katty | it's my cake and i'll nom it if i want to |
| 20:06.08 | miztic | haha fair enough |
| 20:06.15 | miztic | cake does sound good |
| 20:06.16 | JonathanRose | p3nguin: We (elguero and I) are waiting on you to test the latest patch for ASTERISK-18626. Once you've tried it, let us know how it works in the issue. |
| 20:06.31 | zamba | is it possible to transmit sound through usb cable? |
| 20:06.55 | zamba | i'm talking about the chan_mobile solution with usb-connected mobile phone |
| 20:07.29 | WIMPy | zamba: AFAIK no. |
| 20:07.46 | WIMPy | But you can use osmocombb+LCR via USB cable. |
| 20:08.01 | *** join/#asterisk gxdssoft (~gxdssoft@201.230.197.80) |
| 20:10.10 | zamba | WIMPy: what's that? |
| 20:10.48 | WIMPy | it's an alternative firmware for handsets with calypso chipset. |
| 20:11.13 | *** join/#asterisk SpiderMon (~SpiderMon@68.152.22.33) |
| 20:11.20 | zamba | WIMPy: i see - i think :) |
| 20:11.56 | dijib | wow compiling asterisk takes a long time |
| 20:12.03 | WIMPy | Which reminds me that I wanted to look out for such a thing to do exactely that. |
| 20:12.13 | dijib | i just had a 20min shower and she's still not done |
| 20:12.19 | WIMPy | dijib: I used to have such a PC as well. |
| 20:12.28 | Katty | girls always take longer |
| 20:12.29 | WIMPy | Ok, no. Not that bad. |
| 20:12.29 | jaytee | a long time to compile on what kind of system? |
| 20:12.41 | WIMPy | Are you sure you need all the stuff you enabled? |
| 20:12.48 | dijib | heh. its an old p4 laptop i pulled outof a trashbin. |
| 20:12.57 | jaytee | ah, that explains it |
| 20:13.05 | WIMPy | Katty: Don;t believe everything you read in the papers. |
| 20:13.09 | dijib | im absolutly sure i dont need everything installed but ... .migh taswell |
| 20:13.20 | Katty | i don't have to. |
| 20:13.25 | Katty | i know from personal sperience |
| 20:13.55 | jaytee | yeah, older P4s and really old Xeons tend to be slow. Even a D510 Atom takes over 10 minutes to compile. |
| 20:14.21 | WIMPy | My test system is PIII, but that's not THAT bad. |
| 20:14.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:15.13 | _Corey_ | I'm impatient... i always check on how many cpu cores I have and then do a 'make -jX' where X=cores ... :) |
| 20:15.19 | Katty | hi tony |
| 20:15.49 | WIMPy | _Corey_: Are you sure that checking doesn't take too long? |
| 20:16.25 | _Corey_ | WIMPy: checking...? as in 'cat /proc/cpuinfo' |
| 20:16.45 | WIMPy | And what if one of the threads has to wait for the harddisc? It will be idle then. |
| 20:17.43 | _Corey_ | ah... it seems to work out ok. don't ask me how :) |
| 20:18.12 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 20:18.36 | WIMPy | If I am impatient I'd start N+1 to 2*N threads. |
| 20:18.51 | Katty | then i drag you away from your computer |
| 20:18.54 | Katty | and make you drink bear |
| 20:18.57 | Katty | ...beer |
| 20:19.12 | thehar | BEER |
| 20:19.16 | thehar | goes back to lurking |
| 20:19.21 | Katty | *hee* |
| 20:19.25 | _Corey_ | lol, If I have beer I don't care about it going fast :) |
| 20:20.44 | Katty | thehar: you had schafly pumpkin ale yet? |
| 20:20.48 | Katty | thehar: 8% ^________________^ |
| 20:20.49 | thehar | newps |
| 20:20.53 | thehar | sounds taassttty |
| 20:20.59 | Katty | vera-nom-able |
| 20:21.04 | Katty | woah. |
| 20:21.08 | Katty | why wasn't THAT my derby name |
| 20:21.10 | Katty | Vera Nomable |
| 20:21.13 | thehar | lol |
| 20:21.15 | Katty | facepalms |
| 20:22.05 | *** join/#asterisk trumee (~trumee@cpc2-cmbg7-0-0-cust855.5-4.cable.virginmedia.com) |
| 20:22.08 | *** join/#asterisk kinko (~kinko@77.208.134.54) |
| 20:22.13 | kinko | hello guys :) |
| 20:22.17 | dijib | make[2]: *** No rule to make target `anaFilter.o', needed by `libilbc.a'. Stop. |
| 20:22.17 | dijib | make[1]: *** [ilbc/libilbc.a] Error 2 |
| 20:22.20 | Katty | hi kinko |
| 20:22.23 | trumee | hello everybody |
| 20:22.33 | Katty | hi trumee |
| 20:22.36 | kinko | Katty hello girls too :) |
| 20:22.45 | Katty | woo |
| 20:22.48 | *** join/#asterisk talntid (~kbrooker@c-67-168-115-132.hsd1.wa.comcast.net) |
| 20:22.55 | trumee | does anybody have experience of setting up TLS/SRTP on grandstream ATA HT 503? |
| 20:23.30 | *** join/#asterisk xpot-mobile (~james@155-99-213-17.uconnect.utah.edu) |
| 20:23.51 | trumee | I tried to put certificate/private key in grandstream, but it didnt register |
| 20:24.00 | trumee | wondering what i might have done wrong |
| 20:24.07 | Katty | have you tried turning it off and on again |
| 20:24.22 | dijib | need to disable codec |
| 20:24.31 | kinko | I have an easy question today, I would like to add many host=fixed_ip to sip.conf for many peers, and avoid to add one by one, would be fine host=1.2.3.4,1.2.3,5..,1.2.3.200 ? or may be host=somehost.domain.xxx and setup /etc/hosts for many IP's for same domain name ? |
| 20:24.49 | trumee | Katty: yes, i did do that. Let me post the error message |
| 20:25.01 | Katty | trumee: that's a quote from the IT Crowd |
| 20:25.08 | [TK]D-Fender | checkout time, BBL |
| 20:25.08 | Katty | trumee: it was meant to be funny |
| 20:25.16 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 20:25.17 | trumee | Katty: ah right |
| 20:26.14 | nix8n82-phone | Anyone with a cent os 5.5 box know what repository git and jack audio connection kit are in? |
| 20:27.05 | Qwell | wtf. CentOS doesn't have git? |
| 20:27.31 | WIMPy | Qwell: That seems to be the new trend. |
| 20:27.41 | Qwell | it sure doesn't.. I'll be damned. |
| 20:27.50 | WIMPy | The mainstream distributions don't even include the neccessary stuff any more. |
| 20:27.59 | Katty | Qwell: what's git? |
| 20:28.31 | WIMPy | Katty doesn't like fresh meat? |
| 20:28.53 | Katty | are we talking about rollerderby? |
| 20:28.56 | Katty | or something else? |
| 20:29.05 | WIMPy | Software |
| 20:29.21 | WIMPy | But that tends to stink after some time as well. |
| 20:34.00 | trumee | The error is, "Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure" |
| 20:38.12 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:38.18 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 20:38.34 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 20:43.09 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 20:44.00 | JustinCampbell | Can someone help me with an issue? We have an inbound SIP trunk from an Avaya CS1K, and the trunk goes down randomly and gets into a state where Asterisk rejects all registration attempts with a 401. |
| 20:44.07 | JustinCampbell | [2011-09-27 20:40:05.876] DEBUG[8968] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port. |
| 20:44.08 | JustinCampbell | [2011-09-27 20:40:05.876] DEBUG[8968] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port. |
| 20:44.09 | JustinCampbell | We see this in the logs: |
| 20:45.16 | JustinCampbell | Also: Unable to find key 'cs1k-peer-name' in family 'SIP/Registry' |
| 20:49.09 | Micc | JustinCampbell, how many calls per second? |
| 20:49.36 | JustinCampbell | Micc: actual phone calls? Just a few an hour. |
| 20:49.37 | *** join/#asterisk m_tadeu (~quassel@segredosdavida.com) |
| 20:50.03 | JustinCampbell | Micc: it seems to actually disconnect during lower usage periods, maybe a timeout of some sort |
| 20:51.15 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 20:53.39 | pabelanger | JustinCampbell: what version of Asterisk? |
| 20:53.47 | JustinCampbell | pabelanger: 1.8.6 |
| 20:54.10 | JustinCampbell | pabelanger: 1.8.6.0 |
| 20:54.11 | pabelanger | ~collectdebug |
| 20:54.11 | infobot | i guess collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 20:54.31 | pabelanger | JustinCampbell: ^ collect a SIP debug trace an open a new issue on the tracker |
| 20:54.33 | pabelanger | looks like a bug |
| 20:54.44 | JustinCampbell | pabelanger: ok will do thanks |
| 20:57.49 | *** join/#asterisk JD411 (~JD411@nat/digium/x-ubwhfylugtrjgfgp) |
| 20:58.17 | JustinCampbell | pabelanger: i actually have a full debug where the issues was recreated, but minus the sip packets |
| 20:58.24 | JustinCampbell | pabelanger: but ill redo with sip debug on |
| 20:59.01 | pabelanger | JustinCampbell: yup, that's the stuff we'd need to see |
| 20:59.24 | *** join/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 21:00.45 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 21:01.01 | BenC[UK] | Hi Guys |
| 21:01.05 | BenC[UK] | I've had some users log out of sip |
| 21:01.08 | BenC[UK] | but they're still pinging |
| 21:01.15 | BenC[UK] | they've rebooted their machiens they tell me |
| 21:01.21 | BenC[UK] | and not restartd the sip clients |
| 21:01.30 | BenC[UK] | but still the asterisk servers shows them as being online |
| 21:01.53 | Marquis42 | JustinCampbell: I saw your post over on #asterisk-dev that redirected you here. Want to talk about your CS1k issue? |
| 21:01.56 | BenC[UK] | http://pastebin.ca/2083336 |
| 21:02.35 | BenC[UK] | gcollins is one of them |
| 21:02.48 | JustinCampbell | Marquis42: yes please :) |
| 21:03.01 | Marquis42 | :) |
| 21:03.04 | JustinCampbell | pabelanger: the other side has a wireshark capture for the same time frame |
| 21:03.06 | pabelanger | BenC[UK]: enable sip debug for the peer and see what happens |
| 21:03.29 | Marquis42 | OK, well the first question is yes about sip captures. Also, what version of the software are you running? |
| 21:03.54 | JustinCampbell | 1.8.6 |
| 21:03.59 | JustinCampbell | CS1K i think is the latest |
| 21:04.03 | JustinCampbell | its the Avaya test lab |
| 21:04.17 | Marquis42 | Sorry, I mean tthe CS1K. But I suppose the Asterisk version is important as well. :) |
| 21:04.57 | Marquis42 | OK, well if you don't have direct control over it then a sip trace of a call attempt would be extremely helpful. |
| 21:05.04 | BenC[UK] | pabelanger: typically, as soon as I went to do that the users droppd offline |
| 21:05.28 | BenC[UK] | what do I need to change to adjust the "wait" time after a user signs out |
| 21:05.51 | JustinCampbell | Marquis42: The issue is that the registration disconnects. The CS1K connects to Asterisk as a client, so if the registration drops, Asterisk cannot send calls to the CS1K. |
| 21:06.06 | JustinCampbell | Marquis42: they say the CS1K is 7.5 GA |
| 21:06.49 | Marquis42 | Ah, so they're actually having the system register to Asterisk? |
| 21:07.00 | JustinCampbell | yes |
| 21:07.04 | Marquis42 | I definitely have never had to get that working before. |
| 21:07.10 | JustinCampbell | i think theres a SIP gateway in between |
| 21:08.17 | Marquis42 | Well I'm flying a little blind on the newer version because that's not what we run here, but in the system here there is a SIP redirect server (they call it the NRS) as well as one or more signalling servers with the SIP endpoint role running on it/them. |
| 21:08.38 | JustinCampbell | Marquis42: CS1000 SIP Signaling Gateway |
| 21:11.10 | Marquis42 | Right, that's what I meant by the SIP endpoint role. |
| 21:11.51 | Marquis42 | So, explain in detail what happens. It attempts to register and fails? Or it successfully registers and then falls off at re-registration? |
| 21:12.07 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:16.50 | *** join/#asterisk ruied (~ruied@pa4-84-91-140-68.netvisao.pt) |
| 21:17.17 | *** join/#asterisk BillyFred (~smithbd@128.187.233.147) |
| 21:17.40 | JustinCampbell | Marquis42: The registration is successful initially. After some time, usually overnight when the load is lower, the registration disconnects. There are some errors returned in SIP traces such as 401 unauthorized I think. The CS1K engineers are only able to re-register after rebooting the system. |
| 21:18.00 | JustinCampbell | Marquis42: I should also mention that all of our peers are realtime |
| 21:18.49 | *** join/#asterisk Lars_G (~Lars@unaffiliated/lars-g/x-000001) |
| 21:18.52 | Lars_G | Greetings all. |
| 21:20.05 | Lars_G | Question, is there any doable, budgetarly soft manner in which old hybrid (panasonic) KSUs can be integrated with asterisk, to perform a gradual voip migration? |
| 21:21.10 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:21.12 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 21:21.35 | blizzow | Is it possible to enable the http (AJAM?) interface without restarting the entire asterisk instance? |
| 21:26.12 | ruied | Hi! lets say I have a phone with two BLF's keys blinking EXT_A and EXT_B (they are ringing at remote phones made by EXT_B and EXT_C), is there a way so I can pickup a specific ringing phone (EXT_A or EXT_B) on my phone ? |
| 21:28.01 | *** join/#asterisk xpot-mobile (~james@155-99-213-17.uconnect.utah.edu) |
| 21:28.04 | Lars_G | Ok, I see/guess, I could use e1 for this. hmmm |
| 21:29.56 | Marquis42 | JustinCampbell: Have you tried making the one peer that they are using static (i.e., defined in the config file)? |
| 21:30.19 | JustinCampbell | Marquis42: I have not, but it crossed my mind today after looking through the debug logs |
| 21:30.35 | Marquis42 | I'll be honest and say at this point I'm just generally troubleshooting. I've got a working CS1K setup here that takes several hundred calls a day without issue, but we don't do registration with Asterisk at all. |
| 21:30.53 | JustinCampbell | Marquis42: it just sends calls to the CS1K? |
| 21:30.58 | Marquis42 | Yes |
| 21:31.11 | Marquis42 | Well, and receives calls from the CS1K. |
| 21:32.12 | JustinCampbell | We were trying to avoid static hosts for the CS1K, as theyre connecting to a DNS round-robin of Asterisk servers, and their IP could change |
| 21:32.48 | JustinCampbell | Marquis42: I'm heading home for the night, thanks for your help. You too pabelanger |
| 21:33.05 | Marquis42 | Not a problem |
| 21:34.07 | *** part/#asterisk Marquis42 (~bfhbmw0@65-127-126-34.dia.static.qwest.net) |
| 21:45.21 | ruied | Is there a way so I can direct pickup a blinking BLF key ? |
| 21:47.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:47.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:47.44 | ruied | I can use meetme rooms or pickup group, but my problem is if I have 2 or more ringing extensions, I don't know howto distinguish the one I want to pickup up... |
| 21:47.45 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:50.34 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 21:57.20 | Lars_G | Ok, if anyone here has worked with panasonic PBXs. with a kx-td1232 is there any way in which using the E1 expansion (if I find the thing here), that the panasonic treats asterisk as extensions and not as the pstn? |
| 21:57.23 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 21:57.50 | Lars_G | Since we receive FXS via the fxb from the telco, and getting the lines into an E1 would be a real hassle. |
| 21:59.09 | Lars_G | So, the COs are received directly on the panasonic and I wouldn't really want to move them to an FXO card on the asterisk pbx, since then routing extensions from the analogs on the panasonic to the voip extensions on the asterisk would be hell, woudln't it? |
| 22:00.18 | pdtpatrick | Question .. I'm using jabber on asterisk 1.8 but each time i use JABBER_RECEIVE .. asterisk crashes. .. here's my dialplan |
| 22:00.19 | pdtpatrick | http://pastebin.com/7fb1RwrB |
| 22:00.35 | pdtpatrick | so when u get the text message asking u to type back a response |
| 22:00.39 | pdtpatrick | whatever you send back .. asterisk crashes |
| 22:01.05 | pdtpatrick | however without making the call -- u can send something in the same text to asterisk and jabber picks it up fine |
| 22:01.13 | pdtpatrick | just whenever u use JABBER_RECEIVE it crashes |
| 22:02.06 | *** join/#asterisk navaismo (~navaismo@189.230.118.194) |
| 22:03.51 | *** join/#asterisk sflemming (~stefan@85.183.53.64) |
| 22:04.33 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 22:05.13 | sflemming | Help, my asterisk 1.8.7.0 creates approx. 14GB coredumps per hour. Can someone please help me? |
| 22:06.40 | pabelanger | ~backtrace |
| 22:06.40 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 22:06.45 | pabelanger | sflemming: ^ |
| 22:08.12 | sflemming | pabelanger: Thank you, I will try this. Creating core dumps is no problem. Asterisk is writing several hundred per hour |
| 22:08.33 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 22:09.01 | pabelanger | sflemming: we just need 1 core dump for a back trace |
| 22:09.49 | sflemming | pabelanger: yes, I'm just figuring out if backtrace is in portage (I'm using gentoo) then I will try it |
| 22:15.23 | pdtpatrick | I take it no one here knows has toyed with JABBER_RECEIVED :( |
| 22:15.26 | sflemming | pabelanger: When I understand what is written in your link, I shall submit the coredump to your bug tracker? Or should I get more information with gdb? It's a bit unclear. Is it sufficient to submit a dump from /var/lib/asterisk/coredump or do I need to recompile everything? I use asterisk from portage. |
| 22:16.28 | pabelanger | sflemming: no, don't post the coredump. It is useless for anybody else, run it through gdb and attach the output |
| 22:17.42 | sflemming | pabelanger: Okay, I understand. Will need to install gdb... |
| 22:19.14 | sflemming | is there maybe a bug known with the calendar integration? I have 12 google caldav calendar ins calendar.conf and when I load it asterisk crashes regularly. I searched the issue tracker but found nothing about that. |
| 22:19.55 | citywok | sflemming: i had found a bug using exchange, but it was the library (libneon) * was using tha was actually crashing. |
| 22:20.06 | citywok | which then caused * to seg |
| 22:21.44 | pabelanger | sflemming: possible, I have not see anything recently |
| 22:21.52 | sflemming | I think it might be a synchronization bug when many google calendar instances are generated. When there is only one it works. Tested with asterisk from 1.8.5 to 1.8.7 and all the same. It crashes with same mutex error |
| 22:24.42 | *** join/#asterisk umay (~chris@64.92.218.72) |
| 22:25.37 | sflemming | is there maybe someone who is using caldav with the google calendar and can reproduce it? |
| 22:29.29 | *** part/#asterisk irroot (~irroot@197.108.10.147) |
| 22:30.01 | sflemming | hm, it does not seem to work with gdb. I tried db -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c /var/lib/asterisk/coredump/core-20110927-23555542 > /tmp/backtrace.txt but it gives me /var/lib/asterisk/coredump/core-20110927-23555542: Datei oder Verzeichnis nicht gefunden. |
| 22:30.01 | sflemming | No stack. |
| 22:33.39 | *** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230) |
| 22:34.25 | sflemming | When I start asterisk in gdb I get a asterisk: ath.c:193: _gcry_ath_mutex_lock: Assertion `*lock == ((ath_mutex_t) 0)' failed., is this helpful? |
| 22:36.01 | blizzow | one of my users is complaining that he keeps getting left voicemails every few minutes, sometimes even faster. I can see a bunch of debug info regarding the call (http://pastebin.com/8X3UNuuu) but how can I tell where the VM is originating. Is it possible to tell if it was some IVR route, if people are dialing his number directly, if someone is transferring people straight into his VMbox? |
| 22:43.52 | sflemming | have to reset my network... |
| 22:46.30 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 22:47.05 | darkbasic | hi, is there a t38 gateway patch against 1.8.7? |
| 22:48.39 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:00.38 | pabelanger | darkbasic: talk to irrot |
| 23:00.42 | pabelanger | iroot* |
| 23:11.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:24.12 | dijib | wooooohoooooo |
| 23:24.55 | dijib | silent calls are fixed. |
| 23:25.05 | dijib | confrence is back |
| 23:25.07 | dijib | moh is back |
| 23:25.10 | dijib | vm to email is back |
| 23:25.21 | dijib | fax detect to email is back |
| 23:25.44 | dijib | no lumenvox..... |
| 23:27.27 | *** join/#asterisk cstachris (~chrismylo@202.182.147.82) |
| 23:32.51 | *** join/#asterisk BBM (~kvirc@nltaus.lnk.telstra.net) |
| 23:36.35 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 23:37.42 | p3nguin | DId you ever figure out why there was silence on many calls? |
| 23:43.36 | dijib | didnt figure out why. |
| 23:43.43 | dijib | didnt change the dialplan at all. |
| 23:43.48 | dijib | reinstalled and fixed it |
| 23:44.00 | dijib | also it starts automatically now |
| 23:44.09 | dijib | which is very nice |
| 23:44.10 | p3nguin | It's bad that it wasn't figured out why, but it's good that it's fixed. |
| 23:44.31 | dijib | it must have been a bauched install or something i dont know |
| 23:44.39 | dijib | i want lumenvox asr now |
| 23:44.46 | dijib | need to find an expers |
| 23:44.49 | dijib | expert |
| 23:44.50 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 23:44.56 | dijib | or figure out that dependency |
| 23:45.44 | f2knight | Q: Seems Google and Asterisk are at it again... anyone figure out how to 'Answer' the google voice call this time around? |
| 23:46.19 | p3nguin | Something changed? |
| 23:46.31 | p3nguin | I guess I better call myself to make sure it's working. |
| 23:46.43 | dijib | lol |
| 23:46.43 | dym | evenings |
| 23:46.44 | dym | (: |
| 23:46.51 | dym | mornings |
| 23:46.52 | dijib | nothing changed in the .confs |
| 23:46.52 | dym | or whatever |
| 23:46.55 | dym | wherever you're from |
| 23:46.57 | dijib | check pm p3nguin |
| 23:47.06 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 23:48.05 | dijib | get the pm? |
| 23:48.15 | dijib | i even have directory working brother. |
| 23:48.21 | f2knight | p3nguin, well that depends on how you look at it lol. Outbound seems to be working fine but it no longer is accepting the answer.. answer, wait(1), senddtmf(1) |
| 23:48.22 | p3nguin | Google Voice works here for me. |
| 23:48.33 | p3nguin | inbound for me, that is. |
| 23:48.53 | p3nguin | dijib: Yes, I see it. |
| 23:49.07 | f2knight | p3nguin, was working last week for me too. but not now. It rings... just doesnt accept my answer commands. |
| 23:49.41 | p3nguin | Between the time I said I'd better call myself and the time I said it's working for me, I called myself and it's working for me. |
| 23:49.48 | p3nguin | Today. Now. |
| 23:50.51 | p3nguin | dijib: If that's a DID on VoIP.ms, and if I called your number via VoIP.ms, it shouldn't count against your minutes and it won't cost me anything per minute. |
| 23:51.24 | dijib | really? |
| 23:51.29 | dijib | it is on voipms |
| 23:51.30 | p3nguin | Really. |
| 23:51.38 | p3nguin | Free calls between accounts. |
| 23:51.57 | dijib | well good, im getting my family to all make wrt54g asterisk boxes and get voipms accounts |
| 23:52.26 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:52.28 | p3nguin | They say "between DIDs," but anyone with a brain knows you can't call between two inbound articles. |
| 23:52.40 | dijib | lool |
| 23:52.46 | dijib | anyone with a brain eh |
| 23:53.04 | p3nguin | People who run ITSPs are often not very bright. |
| 23:53.21 | dym | <PROTECTED> |
| 23:53.30 | p3nguin | s/who run ITSPs // |
| 23:53.48 | dym | too late to correct. |
| 23:54.23 | dijib | especially the french |
| 23:54.24 | p3nguin | Anyway, I can call you for free on voipms even if I don't have any DIDs with them at all. |
| 23:54.44 | p3nguin | So they should reword their thing. |
| 23:54.55 | dijib | how? |
| 23:55.00 | dijib | GV? |
| 23:55.05 | p3nguin | By dialing, of course. |
| 23:55.20 | dijib | still costs me if you call the 877 |
| 23:55.23 | p3nguin | We're talking about within voipms... where goes GV come into it? |
| 23:55.36 | dijib | thought you were speaking of it earlier |
| 23:55.44 | p3nguin | It won't cost you anything if I call from my voipms account. |
| 23:55.55 | dijib | understood |
| 23:56.03 | p3nguin | Even if I don't have any DIDs. |
| 23:56.12 | dijib | hey what were the res_fax_digium install procedures? |
| 23:56.12 | p3nguin | Because DIDs have nothing to do with me calling out. |
| 23:56.19 | dym | inter itsp calling only requires an accurate * config |
| 23:56.19 | p3nguin | Install it. |
| 23:56.33 | dym | BLA@itsp |
| 23:56.34 | p3nguin | s/inter itsp// |
| 23:56.34 | dym | etc |
| 23:56.43 | dym | whatever |
| 23:56.50 | p3nguin | Just calling. Not inter itsp calling. |
| 23:56.59 | p3nguin | Calling just requires working configs. |
| 23:57.33 | p3nguin | I guess what I'm trying to say without being understood is that you don't need to have a DID to call outbound. |
| 23:57.41 | p3nguin | DIDs have nothing to do with outbound calling. |
| 23:57.49 | dijib | correct |
| 23:58.01 | dijib | but you need an account with voipms |
| 23:58.10 | dijib | fax is failing now. |
| 23:58.11 | p3nguin | But I don't need to buy any phone numbers. |
| 23:58.40 | dym | dijib: depends |
| 23:58.44 | dym | if you have an online asterisk |
| 23:58.45 | dym | you dont |
| 23:58.54 | p3nguin | I can call you all day long on one of your phone numbers, and it won't cost you or me anything. |
| 23:59.17 | dijib | how? |
| 23:59.29 | p3nguin | (1855.04) <p3nguin> By dialing, of course. |
| 23:59.41 | dym | :D |
| 23:59.43 | dym | LOVELY |
| 23:59.50 | dijib | yeh but we would have to have asterisk boxes connected to eachother |
| 23:59.52 | dijib | no? |
| 23:59.55 | p3nguin | No. |
| 23:59.59 | p3nguin | VoIP.ms does that for us. |
