| 00:01.53 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 00:04.08 | seather | thank you, found a post online, in sangoma configuration needed to disable hardware fax detection |
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| 00:31.58 | dym | where did the jabber support disappear to? :| ive upgraded to 1.8.7.0 |
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| 01:03.27 | ghostmediapro1 | after viewing asterisk log file i keep seeing this link https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
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| 01:09.13 | benklop | did chan_gtalk break again some time recently? |
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| 01:13.23 | puzzled | benklop: yes they changed something again. this patch should fix it: https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 01:26.39 | benklop | puzzled: ah, that one again. |
| 01:27.03 | puzzled | yup, the can't seem to make up their mind about that one |
| 01:29.27 | benklop | is there a reason the code is searching for redirect or sta:redirect, and not just "some text ending with/containing redirect" or something else slightly more intelligent than an exact match? |
| 01:30.38 | benklop | i don't know enough about the protocol to know what is actually allowed in the response here |
| 01:30.47 | puzzled | neither do I |
| 01:32.20 | benklop | do you know how long it has been broken this time? last time they reverted the change a few hours after I applied the patch |
| 01:32.22 | benklop | :-P |
| 01:35.14 | puzzled | no idea |
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| 03:56.19 | adeel | sorry for being ignorant on the matter; but is there a reason why voicemail storage in a database must be through ODBC? |
| 03:57.49 | adeel | why can't it be stored directly to a mysql/psql db? i'm aware that not all databases handle their blobs the same way...but isn't that what the different res_ modules are for? |
| 03:59.02 | [TK]D-Fender | adeel: They didn't want to code for just one engine. Why waste non-recyclable code? Just use MySQL via a DSN. |
| 03:59.31 | adeel | [TK]D-Fender: then why not just have all db communications via odbc? |
| 03:59.51 | *** join/#asterisk samuelsapps (~samuel@202.137.7.242) |
| 04:00.09 | [TK]D-Fender | Also direct integration is more necessary for run-time things for performance reasons. VM wouldn't get hit as much as dialplan, cdr, etc |
| 04:00.39 | adeel | hmmm...fair enough |
| 04:00.47 | [TK]D-Fender | [23:59]adeel[TK]D-Fender: then why not just have all db communications via odbc? <- I agree. Some things might deserve integration performance, and there is "established code" |
| 04:01.13 | [TK]D-Fender | We've got what we already have... so I don't think they will be too eager to just pull it. |
| 04:01.26 | adeel | i assumed as much; but just wanted verification |
| 04:01.39 | [TK]D-Fender | However "agnostic" should be the goal of * where possible one would think. |
| 04:02.16 | adeel | i wonder what the performance penalty would be if you compared ODBC vs Direct |
| 04:03.32 | [TK]D-Fender | adeel: CDR should be so bad; how fast would those collect? Guess dialplan & SIP would be the big hits (imagine 1000 peers on qualify), lots of call processing, etc... |
| 04:03.38 | [TK]D-Fender | adeel: that could add up |
| 04:03.49 | [TK]D-Fender | oops CDR = no bad |
| 04:04.21 | adeel | [TK]D-Fender: but those could be cached or optimized |
| 04:04.53 | [TK]D-Fender | Well I don't know the fine points... anyway if you're more curious you could always post up on the ML's |
| 04:05.01 | SeRi | [TK]D-Fender, quick question. What usually generates this errors? pbx_load_config: Can't use 'next' priority on the first entry is it sip.conf or extensions.conf? |
| 04:05.33 | adeel | besides, can a single instance handle 1000 peers or so? i haven't kept up with * performance since 1.4 |
| 04:05.42 | *** join/#asterisk radic (~radic@dslb-178-002-222-231.pools.arcor-ip.net) |
| 04:06.14 | [TK]D-Fender | serjust like it sounds.. you have an "n" without a "1" |
| 04:06.21 | SeRi | a shit |
| 04:06.22 | SeRi | thanks |
| 04:06.24 | SeRi | you are right |
| 04:06.29 | SeRi | :P |
| 04:06.47 | [TK]D-Fender | SeRi: You mean "it" is right. Just like it is saying it is... |
| 04:07.37 | [TK]D-Fender | adeel: They've managed to push a single box past 10000 calls (no media), so yes, 1000 isn't too bad depending on trancoding, recording, etc |
| 04:08.05 | adeel | [TK]D-Fender: must have been a pretty beefy box |
| 04:08.50 | [TK]D-Fender | adeel: recording I never took as being a "real" load as you could mount an SSD or RAM disk as a drop-zone for the real-time dump, and hard-copy at the end. since al sequential would be for storage even that wouldn't be a real "hit" |
| 04:08.58 | [TK]D-Fender | adeel: that was 2-3 years ago |
| 04:10.38 | adeel | [TK]D-Fender: i'm still sure that number will drop once you add in media (even if there is no transcoding at all)... |
| 04:11.27 | adeel | [TK]D-Fender: and i agree with you're take on recording...it doesn't really require much |
| 04:11.38 | adeel | er, your, not you're |
| 04:11.56 | Naikrovek | odbc doesn't add much overhead at all. |
| 04:11.58 | [TK]D-Fender | adeel: It could only drop, but how much for just media? anything where added disk IO comes in the the big one (and trancode) |
| 04:12.57 | adeel | [TK]D-Fender: yeah, that's the interesting question..maybe i'll get some free time and setup a test environment and try testing it out in a few weeks |
| 04:14.18 | adeel | Naikrovek: i wouldn't expect it to add much; but you never really know |
| 04:14.43 | Naikrovek | well it's easy to benchmark. it doesn't add much overhead at all. |
| 04:14.53 | Naikrovek | at least not when I tested it a few years back. |
| 04:16.28 | [TK]D-Fender | Media being repacked isn't a big deal... |
| 04:16.51 | [TK]D-Fender | Strip header, new header. Transmit. |
| 04:17.00 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 04:17.02 | Naikrovek | the bottleneck for asterisk server performance will be network IO and CPU use for mixing audio. |
| 04:17.27 | Naikrovek | both can be mitigated to a degree |
| 04:17.47 | Naikrovek | heads to slumberland |
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| 04:31.20 | SeRi | [TK]D-Fender, Is it possible to setup an extension to use a prefix to dial out using a different account? IE: primary account is voip.ms but I want the user to dial *22 + local number and be able to use the secondary provider... Is it possible? |
| 04:34.15 | [TK]D-Fender | SeRi: Of course. Your dialplan will do whatever you make it do |
| 04:34.55 | SeRi | Mhhh ok I am ready and trying to figure out how too. Thanks for confirmation. Any pointers? |
| 04:35.13 | [TK]D-Fender | serYour 2nd dialplan pattern is almost no diffferent from your first. |
| 04:35.37 | [TK]D-Fender | SeRi: start the pattern with *22 and dial out the other provider. That's all there is. |
| 04:35.56 | SeRi | got it. Thanks |
| 05:03.01 | SeRi | [TK]D-Fender, exten => _1NXXNXXXXXX |
| 05:03.09 | SeRi | do I add *22 in front of it? |
| 05:03.53 | SeRi | mhh myabe *22| |
| 05:05.55 | p3nguin | How are you going to dial the vertical bar symbol? |
| 05:06.51 | p3nguin | Do you want to have to dial the *22 before the 1? |
| 05:07.44 | SeRi | yes |
| 05:07.51 | SeRi | to define which route to use |
| 05:07.59 | p3nguin | Then put the *22 before the 1. |
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| 05:12.41 | SeRi | I did that and it does not work exten => _*221NXXNXXXXXX |
| 05:13.03 | SeRi | It ttrys to dial out and I get a "could not complete your call" |
| 05:13.06 | p3nguin | That is the correct extension pattern. |
| 05:13.14 | p3nguin | But... |
| 05:13.18 | *** part/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 05:13.41 | p3nguin | You're probably trying to dial ${EXTEN}, which is the entire extension you just entered. |
| 05:14.17 | p3nguin | So you need to offset your exten variable by 3 to not pass the *22 along to the provider. |
| 05:14.38 | SeRi | exten => _*221NXXNXXXXXX,n,Dial(SIP/sipbri/${EXTEN}) |
| 05:14.52 | p3nguin | That's why it doesn't work. |
| 05:15.21 | p3nguin | If you dialed *223145551212 on your phone, ${EXTEN} is going to be *223145551212 |
| 05:15.28 | p3nguin | And they won't like that phone number.] |
| 05:15.42 | SeRi | Ok I see |
| 05:15.48 | p3nguin | Offset the variable. |
| 05:19.03 | SeRi | p3nguin, exten => _*221NXXNXXXXXX,n,Dial(SIP/sipbri/${EXTEN}:0:3) |
| 05:19.24 | p3nguin | probably :3 not :0:3 |
| 05:19.42 | p3nguin | ${EXTEN:3} |
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| 05:19.59 | SeRi | ooo yes you are right |
| 05:20.35 | p3nguin | :0:3 would offset it 0 and use only 3. |
| 05:22.54 | SeRi | Thats right |
| 05:24.06 | SeRi | ok now all I get its a busy tone. |
| 05:26.45 | SeRi | Mhhh something is not right |
| 05:26.54 | SeRi | is not striping the digits as told |
| 05:27.19 | SeRi | To: <sip:*221MYTEL:3@sip2.remacservices.net> |
| 05:31.22 | [TK]D-Fender | seryou put the :3 after the } instead of inside |
| 05:31.43 | SeRi | lol I just caught that! |
| 05:31.47 | SeRi | you beat me to it :P |
| 05:32.28 | SeRi | I think I been told by you a few times to get some rest and not play around with stuff when tired lol |
| 05:33.04 | SeRi | ok guys I am off to bed. Ill deal with this when rested |
| 05:33.17 | SeRi | Thank you [TK]D-Fender and p3nguin |
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| 05:51.48 | schmidts | good morning |
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| 07:28.48 | sehh | hey people |
| 07:32.06 | sehh | q: I'd appreciate some help. I'm using asterisk 1.6 on an old server. I'd like to create a feature code for sending a sequence of DTMF digits to the phone line. For example, when I pick up the phone, I'd like to dial "*90" which would send "*30*55555#" to the line and then hung up. Is this possible? |
| 07:32.51 | tuxx- | sehh: the dial() application has a parameter to send dtmf digits, if thats what you mean. |
| 07:34.14 | WIMPy | Sounds more like the quick dial thing to me. But it isn't exactely clear. |
| 07:34.25 | sehh | I also found the SendDTMF() which does the same, I believe the Dial() takes a "D" parameter for sending DTMF to the line |
| 07:34.30 | kaldemar | sehh: do you mean it should just dial *30*55555#? when should it hang up? |
| 07:34.40 | sehh | kaldemar, thats correct |
| 07:34.54 | kaldemar | sehh: when should it hang up? |
| 07:35.05 | sehh | once its done dialing, it doesn't have to wait |
| 07:35.40 | kaldemar | set a timeout and hope it works. |
| 07:36.00 | sehh | unfortunately, I don't know how to do any of this |
| 07:36.07 | kaldemar | D() only sends DTMF once the channel receives an answer. |
| 07:36.12 | sehh | I'm looking for some newbie help please |
| 07:36.38 | kaldemar | sehh: what kind of a channel are you using? DAHDI? |
| 07:37.09 | sehh | I'm using an ISDN phone line, connected to a BeroNET ISDN card |
| 07:37.18 | sehh | so no DAHDI for me |
| 07:37.30 | kaldemar | what is it then? |
| 07:37.36 | WIMPy | Oh, you will probably have to send that as keypad then. |
| 07:37.42 | kaldemar | i.e. how do you use the card from asterisk? |
| 07:37.55 | WIMPy | But I'm not sure that's possible with Asterisk. |
| 07:37.56 | sehh | mISDN is the interface for the card |
| 07:39.09 | WIMPy | Can you dial that code manually? |
| 07:39.30 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 07:39.37 | sehh | well, only if I remove the server and hook up a normal phone, then yes |
| 07:40.04 | sehh | but not via asterisk, because asterisk tries to interpret the DTMF and never sends anything to the line |
| 07:40.29 | WIMPy | It's probaly exactely the opposite. |
| 07:40.44 | sehh | what do you mean? |
| 07:41.00 | WIMPy | Oh, you will probably have to send that as keypad then. |
| 07:42.13 | WIMPy | Ok, google says it might work. |
| 07:42.52 | WIMPy | You set a varable _MISDN_KEYPAD and then dial nothing. |
| 07:44.05 | WIMPy | exten => *90,1,Set(_MISDN_KEYPAD=*30*55555#) |
| 07:44.10 | WIMPy | exten => *90,n,Dial(misdn/1/) |
| 07:44.17 | sehh | hmm I'm reading something like that in the misdn.org FAQ section |
| 07:45.11 | WIMPy | Or on misdn2 you just dial with /k. |
| 07:45.33 | WIMPy | But good to know. That never crossed my way. |
| 07:45.39 | sehh | looks like this is exactly what I need |
| 07:46.08 | sehh | the FAQ doesn't explain how to use this dialplan |
| 07:46.23 | sehh | sorry I'm a newbie and I am not sure what I need to do |
| 07:46.49 | WIMPy | I wrote it for what you asked in the beginning. |
| 07:47.30 | sehh | ah sorry I missed that |
| 07:48.46 | sehh | thank you very much for the help, much appreciated |
| 07:48.53 | sehh | I'm going to test this right now |
| 07:49.52 | sehh | interestingly, the FAQ uses {$TARGET} |
| 07:49.53 | WIMPy | Wasn't it nice when there used to be a divertctl command? |
| 07:50.34 | sehh | I wouldn't even know, I setup this asterisk server several years ago and have't touched it since, I've forgotten all about it. Quite reliable! |
| 07:50.38 | WIMPy | The example to for setting up forwarding to some {target}. |
| 07:51.18 | sehh | how would it set/get the target variable? is there an extra step? the FAQ doesn't explain |
| 07:51.29 | WIMPy | Yes |
| 07:53.34 | WIMPy | makes a note to see if divertctrl could be ported to misdn. |
| 07:53.42 | WIMPy | That might be quite usefull. |
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| 08:48.16 | ironm | good morning. Can you point me to asterisk-1.6.2.20 rpm packages for CentOS 5.7, please? Thank you in advance. |
| 08:48.51 | ironm | If there are no such a .spec file for CentOS 5.7 would be also helpful. |
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| 09:42.17 | eject_ck | hi all, how it works if I have phone with 2 sip registrations on asterisk and want to make conference with 2 other peers but without using call to 900 ? |
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| 09:43.52 | irroot | eject_ck that is a 3 party conf |
| 09:44.14 | irroot | many phones support this via the conf button on the phone/device |
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| 09:52.42 | Uatec[work] | hi there |
| 09:53.03 | Uatec[work] | i have a device in my asterisk server which lspci identifies as a Tiger Jet Inc, Tiger3xx |
| 09:53.20 | Uatec[work] | that, however is not the name of the device (maybe just the chipset) |
| 09:53.34 | Uatec[work] | how can i find, remotely, the name of the device i have? |
| 10:07.00 | ironm | Hello. Can you point me to asterisk-1.6.2.20 rpm packages for CentOS 5.7, please? Thank you in advance. |
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| 10:16.21 | puzzled | ironm: if you mean the Digium provided rpm packages, check www.asterisk.org. If they are not there then they have not been created (yet) |
| 10:17.28 | ironm | thank you very much for your hint puzzled |
| 10:22.05 | aberrios | anyone know of a free softphone that has a attended-xfer button? |
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| 10:52.19 | din3sh | hi all |
| 10:52.38 | din3sh | Am having call hangups with these messages |
| 10:52.39 | din3sh | [Oct 11 14:23:55] DEBUG[1159] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on Zap/2-1 [Oct 11 14:23:55] DEBUG[1159] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Oct 11 14:23:55] DEBUG[1159] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 |
| 10:53.03 | din3sh | any idea why this occuring? |
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| 11:49.38 | ollii | just a question in asterisk 1.8 + dahdi 2.5 ... if i connect a dahdi bri card (like hfc) to a ntba with 2 ports... what happens to a call if it does not find an approriate extensions? will it get back to the second port or is there something like sangoma has : http://wiki.sangoma.com/BriAdvancedOptions#msn |
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| 12:19.07 | din3sh | Anyone ever come across this? ----> chan_dahdi.c: Not yet hungup... Calling hangup once with icause |
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| 12:35.23 | _jdccdevel | Hello! I'm experiencing a strange intermittent problem with my asterisk voicemail. Every so often, a voicemail message gets stuck in the system. I have a user right now that has Two voicemails stuck in their mailbox, and I'm hoping someone here can help me figure out whats going on. |
| 12:36.51 | _jdccdevel | Asterisk version: 1.6.2.16.2 Voicemail is using on-disk storage. Users voicemail directory looks like this: |
| 12:37.14 | _jdccdevel | root@zodiac:/var/spool/asterisk/voicemail/default/XXXYYYZZZZ/INBOX# ls -l |
| 12:37.16 | _jdccdevel | total 1252 |
| 12:37.17 | _jdccdevel | -rw------- 1 asterisk asterisk 59631 2011-10-09 10:17 msg0003.gsm |
| 12:37.19 | _jdccdevel | -rw------- 1 asterisk asterisk 283 2011-10-09 10:17 msg0003.txt |
| 12:37.20 | _jdccdevel | -rw------- 1 asterisk asterisk 578284 2011-10-09 10:17 msg0003.wav |
| 12:37.22 | _jdccdevel | -rw------- 1 asterisk asterisk 58755 2011-10-09 10:17 msg0003.WAV |
| 12:37.23 | _jdccdevel | -rw------- 1 asterisk asterisk 48246 2011-10-09 10:17 msg0004.gsm |
| 12:37.25 | _jdccdevel | -rw------- 1 asterisk asterisk 287 2011-10-09 10:17 msg0004.txt |
| 12:37.26 | _jdccdevel | -rw------- 1 asterisk asterisk 467884 2011-10-09 10:17 msg0004.wav |
| 12:37.27 | *** kick/#asterisk [_jdccdevel!~pabelange@asterisk/contributor-and-bug-marshal/pabelanger] by pabelanger (_jdccdevel) |
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| 12:37.56 | pabelanger | _jdccdevel: please use pastebin |
| 12:38.14 | _jdccdevel | Sorry, I didn't know my IRC client was going to do that. |
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| 12:39.28 | ollii | <PROTECTED> |
| 12:39.32 | _jdccdevel | http://pastebin.com/sFyXYec4 |
| 12:39.34 | ollii | _jdccdevel: ~b |
| 12:39.36 | ollii | ~pb |
| 12:39.36 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 12:39.44 | ollii | sry to late |
| 12:40.44 | _jdccdevel | Sorry for the flood. I've never had to paste anything to a IRC channel before. |
| 12:41.44 | *** join/#asterisk sehh (~sehh@cust-224-67.on1.ontelecoms.gr) |
| 12:41.47 | sehh | hey people |
| 12:41.48 | sehh | I'm back |
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| 12:42.11 | _jdccdevel | I think the problem is that somehow the messages aren't being properly detected (i.e. they are named msg0003 and msg0004 when "comedian Mail" is expecting them to be msg0000 and msg0001 ? |
| 12:43.46 | sehh | earlier today, I was asking for help with sending DTMF to the phone line, in order to enable/disable various features provided by my phone provider. I'm using ISDN lines with mISDN (chan_misdn). We found that its possible to do that with _MISDN_KEYPAD. Unfortunately, while this seems to work for Germany, it doesn't for me. The phone provider says that I've sent an invalid command. |
| 12:43.52 | sehh | any help would be appreciated please |
| 12:45.34 | [TK]D-Fender | _jdccdevel, check the other folders. |
| 12:48.20 | _jdccdevel | All the other folders have messages starting at 0000 (There aren't that many though, it's a small system) |
| 12:49.10 | [TK]D-Fender | _jdccdevel, Since you have 2 stuck... I'm betting its the OLD bmessages... |
| 12:50.16 | _jdccdevel | "OLD bmessages" ? If you mean the messages are old, they're from october 9th according to the timestamp. |
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| 12:57.56 | [TK]D-Fender | _jdccdevel, No, not the INBOX, the OLD folder |
| 12:58.18 | [TK]D-Fender | _jdccdevel, because * shouldn't renumber VM's IIRC |
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| 13:08.00 | _jdccdevel | [TK], the "Old" folder is empty. |
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| 13:11.30 | _jdccdevel | Another thing I forgot to mention, I cannot hear either voicemail. it just skips over them like they're empty. When I do an asterisk trace, it doesn't even show it trying to play the voicemail. |
| 13:12.54 | Katty | cries. |
| 13:13.36 | irroot | yo Katty what up |
| 13:14.02 | Katty | ear infection HURTS |
| 13:15.38 | [TK]D-Fender | _jdccdevel, PB the folders (all of them) and a call to VoiceMailMain. Walk through everything |
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| 13:16.42 | irroot | Katty its what happens listening to users crap ... |
| 13:19.56 | _jdccdevel | http://pastebin.com/S7yfRMeu |
| 13:27.18 | _jdccdevel | http://pastebin.com/090Sx6EA |
| 13:27.37 | _jdccdevel | The mailbox with the stuck messages is "XXXYYY3427" |
| 13:32.05 | _jdccdevel | [TK], if * doesn't rename (renumber) voicemail messages, is there a "map" or something maintained somewhere that could have got out of sync? |
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| 13:37.21 | _jdccdevel | Interesting! I renamed one of sets of files in the voicemail, and it now lets me hear that voicemail. The voicemail trace is here: http://pastebin.com/kWFqVdu7] |
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| 13:54.43 | as001 | exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)}) |
| 13:54.54 | as001 | That I get from cli with asterisk 1.8.7.0 |
| 13:55.14 | as001 | <PROTECTED> |
| 13:55.23 | as001 | do you know what is this all about ? |
| 13:55.56 | [TK]D-Fender | as001, "core show function CHANNEL". PB it |
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| 13:56.32 | Orbixx | Which version of Asterisk is asterisk-2.1.0-rc1 targeted to work with? |
| 13:56.35 | Orbixx | er |
| 13:56.41 | Orbixx | asterisk-gui-2.1.0-rc1 rather |
| 13:58.17 | [TK]D-Fender | Orbixx, 1.6.x & 1.8 I'm betting |
| 13:58.19 | serafie | Orbixx: it is tested with 1.4-1.8 |
| 13:58.39 | serafie | it *should* work with 10, but I have not tested it. |
| 13:59.15 | as001 | I read it it seems I used proper syntax |
| 13:59.42 | Orbixx | Currently on 1.6.2.2 and for the life of me I cannot get the SIP stack on Android to register with my Asterisk installation, despite my Linksys SPA941 being able to without issue. |
| 14:00.07 | Orbixx | (not using a GUI either, the prior question was mostly curiosity) |
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| 14:00.37 | Orbixx | Anybody here tried using the Android SIP stack with Asterisk? |
| 14:01.11 | Orbixx | I'm betting on a no :< |
| 14:01.40 | [TK]D-Fender | Orbixx, before jumping to the conclusion that it is app related, how about pastebin-ing SIP debug for your register attempts and yoru peer to match masking only passwords... |
| 14:01.42 | Orbixx | Has the config syntax changed much since 1.6.2.2? |
| 14:01.49 | Orbixx | OH |
| 14:01.51 | Orbixx | what an idiot |
| 14:01.55 | Orbixx | I forgot about the sip debug option |
| 14:01.56 | as001 | it says rtpqos takes 2 arguments i put audio and local_lostapckets don't see any mistake |
| 14:02.05 | Orbixx | [TK]D-Fender: Thanks for the reminder |
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| 14:02.18 | [TK]D-Fender | Orbixx, "Oh wait, I should look, shouldn't I?" :) |
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| 14:02.37 | Orbixx | [TK]D-Fender: I was actually tcpdumping earlier |
| 14:02.47 | [TK]D-Fender | Orbixx, always work internal. |
| 14:02.51 | Orbixx | But sip set debug seems more sensible |
| 14:06.22 | WIMPy | Ooops. VoiceMail just told me I had 3 new messages, but played only 2 then saying "no more messages". MWI stayed on and on the next call I got the last message, which was in fact either the 1st or |
| 14:06.26 | WIMPy | 2nd. |
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| 14:09.05 | as001 | [TK]D-Fender can you give me an example of CHANNEL rtpqos ? Is my syntax ok ? |
| 14:09.29 | irroot | WIMPy freaky ... ps you use mISDN in kernel or standalone looks like standalone is a winner ? |
| 14:09.35 | [TK]D-Fender | as001, I'm not seeing a pastebin... |
| 14:11.12 | Orbixx | [TK]D-Fender: Layer 8 error |
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| 14:11.38 | [TK]D-Fender | Orbixx, OSI -> OIC |
| 14:11.40 | as001 | <PROTECTED> |
| 14:11.55 | Orbixx | ;) Thanks for the push in the right direction |
| 14:12.16 | _jdccdevel | [TK]D-Fender, Thanks for looking at my voicemail issue. unless you have any other suggestions, I'm going to try renaming the messages to 0000 and 0001 and let the user try to delete them |
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| 14:14.25 | [TK]D-Fender | Orbixx, You're welcome. |
| 14:16.16 | [TK]D-Fender | as001, Yeah, it looks sane, but the syntax doesn't clearly show the format for delimiters, perhaps it isn't standard as we think it should be... |
| 14:16.51 | as001 | do you suggest that I try with () for arguments ? |
| 14:17.16 | as001 | like this ? exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos(audio,local_lostpackets))}) ? |
| 14:18.16 | [TK]D-Fender | as001, test other pars, and other multi-part parms |
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| 15:12.51 | arnotixe | hi all I've got two asterisk servers set up, the main on a dyndns address. The other one mobile, and registering to the main with iax2. However, when I "iax2 show peers" on the main, it looks ok (remote ip, status OK 208ms). But on the mobile, it says (null) for host and UNKNOWN for status. Is this normal? |
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| 15:13.34 | WIMPy | 2nd. |
| 15:13.54 | [sr] | hi WIMPy |
| 15:15.55 | WIMPy | arnotixe: Do yu have qualify enabled? |
| 15:16.00 | WIMPy | Hi [sr] |
| 15:18.17 | arnotixe | yes, on both sides. |
| 15:18.18 | arnotixe | hmm reading http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers it seems IAX is kind of one-way, so that I will have to create incoming trunks for both servers? |
| 15:19.30 | WIMPy | No, it's both ways. |
| 15:19.46 | WIMPy | Only one peer needed. |
| 15:19.55 | arnotixe | hmm ok here's pasted: http://pastebin.com/mvr9eDTW |
| 15:20.08 | arnotixe | interestingly, only one side thinks it's connected... |
| 15:21.17 | WIMPy | You have different peers? |
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| 15:21.25 | arnotixe | hm? |
| 15:21.50 | WIMPy | I don;t find matching peer names in those two outputs. |
| 15:22.44 | nephfl | is there a good opensource(free) way to check MOS score for an asterisk system? |
| 15:29.37 | arnotixe | WIMPy, hm I might have been confused on the peer names. But I corrected them now, however still only one side thinks the connection is up. New paste including config files: http://pastebin.com/c9Z8qBST contains |
| 15:30.57 | arnotixe | iax2 set debug on on the REMOTE actually shows some traffic coming and going, though. |
| 15:35.14 | WIMPy | You should put the host name instead of dynamic on REMOTE. |
| 15:35.45 | WIMPy | With dynamic it can't know where to find MAIN. |
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| 15:36.17 | WIMPy | Register and peer both need the correct information. |
| 15:36.49 | [TK]D-Fender | ^^^ |
| 15:38.09 | arnotixe | ah so in the corresponding section of iax.conf it needs my host name on both places? I'll try that. |
| 15:38.58 | arnotixe | now that's different thank you! |
| 15:39.26 | [TK]D-Fender | arnotixe, the side that registeres knows where the other side is. it needs that host in its peer |
| 15:39.54 | [TK]D-Fender | arnotixe, the side that you are registering to needs to know where the other side is and does so because they register |
| 15:40.04 | arnotixe | [TK]D-Fender, yep. I thought host=was to specify the IP, and register was to specify host name |
| 15:40.17 | arnotixe | Now I get on the main "Rejected connect attempt from 190.131.185.61". But it's progress :D |
| 15:40.34 | [TK]D-Fender | arnotixe, No, register means tell the other side's peer where you are so they can call you |
| 15:42.05 | WIMPy | No, on MAIN you keep the host=dynamic. |
| 15:42.14 | WIMPy | Otherwise it won't accept registrations. |
| 15:42.15 | arnotixe | ah that makes sense if I thought like a computer :D |
| 15:42.35 | arnotixe | yes, on main it's host=dynamic. |
| 15:42.35 | WIMPy | registering is there to tell the other side where to reach you. |
| 15:42.43 | WIMPy | So either you configure the host or you register. |
| 15:42.54 | arnotixe | I think the reject is some dialplan problem: "[Oct 13 17:39:20] NOTICE[31050]: chan_iax2.c:10062 socket_process: Rejected connect attempt from 190.131.185.61, who was trying to reach '9190509507@'" |
| 15:43.04 | arnotixe | I see there's nothing after the @.. |
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| 15:43.31 | WIMPy | Or you configure a context in the peer definition. |
| 15:44.40 | arnotixe | good idea. For now I'm receiving in the debug on the remote: REJECTED: No authority. Both sides now display OK for iax2 show peers |
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| 15:58.06 | arnotixe | ok the iax trunk seems to be connected. However I get rejected calls, IAX reject cause 50 (No authority found). Does both servers need to know about each other's peers? |
| 15:59.36 | arnotixe | http://pastebin.com/FUGpaBFr is what debug looks like on the remote server |
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| 16:29.40 | Micc | Is thre a character my customers can program into a speed dial key on a polycom to pause for a bit? |
| 16:30.10 | Micc | Google seems to be only showing things to put into the config file. |
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| 17:12.08 | *** join/#asterisk docid (~PISSS@69.17.142.138) |
| 17:12.39 | docid | hello, trying to figgure out how to catch calls with no callerid and replace it with something generic, while leaving the calls with proper callerid intact |
| 17:13.14 | docid | asterisk 1.4.42 |
| 17:13.17 | [TK]D-Fender | Check if callerid is blank. if not, skip ahead. |
| 17:13.24 | [TK]D-Fender | if so, change it |
| 17:13.36 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:13.46 | [TK]D-Fender | "show function CALLERID" |
| 17:13.49 | docid | yes, thats what i want to do |
| 17:13.51 | [TK]D-Fender | "show application GOTOIF" |
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| 17:17.45 | docid | thanks, should help |
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| 17:24.54 | mhaddog | afternoon... |
| 17:25.12 | mhaddog | I'm having issues playing custom recordings and MOH under asterisk 1.8.7.0 |
| 17:25.30 | mhaddog | I know the recordings are in the correct format... and the MOH I'm using the default ones.... |
| 17:26.13 | mhaddog | the file seems to be played, thus no error is showed, and the caller just hear silence.... |
| 17:26.18 | mhaddog | any ideas? |
| 17:26.58 | mhaddog | besides ticked number 18262 i haven't foudn something more relevant refering to this... |
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| 17:44.00 | docid | [TK]D-Fender, ok, so heres the problem, how do i detect its blank? i know how to match data there, but so like, GOTOIF(${CALLERID}="" for blank? |
| 17:44.28 | docid | err, ${CALLERID(num)} |
| 17:45.10 | [TK]D-Fender | docid, every character on either side of the = counts. " chars are literal and need to wrap on both sides. Also... read CHANNELVARIABLES.TEX and learn how to use expressions. |
| 17:46.01 | docid | k |
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| 17:48.00 | timholum | Is it possible to set up chan_unistim to work with multiple ports? |
| 18:11.57 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 18:12.34 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 18:12.43 | BenC[UK] | Evening guys |
| 18:13.17 | BenC[UK] | Got a really weird issue... we just lost a carrier.. so I changed our config to take them out of the loop, but for some reason - my failover code to try multiple carriers one after another has stopped working |
| 18:13.40 | BenC[UK] | it appears that the variables I set in extensions.conf in [global] section aren't available in my dialplan |
| 18:15.12 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:15.18 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 18:15.51 | BenC[UK] | extensions.conf: http://pastebin.ca/2089817 dialplan: http://pastebin.ca/2089818 |
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| 18:19.57 | *** part/#asterisk mhaddog (~mhaddog@173-162-115-101-miami.hfc.comcastbusiness.net) |
| 18:21.22 | [TK]D-Fender | BenC[UK], those are constants, not "variables" and are parsed out on dialplan load, not on execution. |
| 18:21.35 | [TK]D-Fender | BenC[UK], Use AstDB or some other means to storing and retrieving those |
| 18:21.59 | BenC[UK] | they have been working fine for months |
| 18:22.15 | BenC[UK] | im happy for them to be done on load |
| 18:22.19 | BenC[UK] | they dont change on the fly |
| 18:25.49 | [TK]D-Fender | exten => s,2,Set(CARRIER=${EVAL(${CARRIER${CALL_ATTEMPT}})}) <--- on the fly |
| 18:25.57 | [TK]D-Fender | You're trying to reverse-parse a constant |
| 18:26.16 | [TK]D-Fender | ${CARRIER${CALL_ATTEMPT}} |
| 18:26.58 | *** join/#asterisk mikemking (~mikemking@static-108-16-123-66.phlapa.fios.verizon.net) |
| 18:27.18 | BenC[UK] | its sorted |
| 18:27.24 | BenC[UK] | I had [global] and not [globals] |
| 18:27.39 | BenC[UK] | yes sorry, I meant the values fo CARRIER1,2,3,4 didnt change |
| 18:27.44 | mikemking | looking for some help doing IP based authentication for peers/users |
| 18:27.46 | BenC[UK] | the eval works fine |
| 18:28.00 | mikemking | specifically, I'm populating permit/deny and it seems to have no effect |
| 18:31.29 | p3nguin | r0m|u, seri: Still didn't show up. |
| 18:31.42 | r0m|u | wtf? |
| 18:32.05 | p3nguin | How did you send it this time? First class again, or priority? |
| 18:32.09 | r0m|u | are you serious? |
| 18:32.43 | p3nguin | Is there any way to track it? |
| 18:33.14 | r0m|u | first class. As priority was going to be a bit over kill as you stated. I talked to the damn Post Office here at school about it |
| 18:33.38 | r0m|u | they said that sims should not be held up any post office |
| 18:34.00 | r0m|u | they are nether dangerous or suspicious items |
| 18:34.10 | p3nguin | I would expect it to take not less than three days for first class. |
| 18:34.19 | p3nguin | This is only the third day. |
| 18:34.48 | p3nguin | Of it you don't count Tuesday, this is the second day. |
| 18:35.16 | p3nguin | Maybe tomorrow or Saturday I'll get it. |
| 18:35.23 | r0m|u | If the sim is not there by tomorrow I will request an investigation at the PO here at school because that is BS! Not damn possible! |
| 18:35.40 | p3nguin | Did you send two of them through the school? |
| 18:35.57 | r0m|u | yes both. the tmobile and the generic sim |
| 18:36.16 | p3nguin | Any chance they got misplaced instead of being mailed out? |
| 18:36.44 | r0m|u | I doubt it. Maybe missed placed in somebodys damn pocket |
| 18:36.58 | r0m|u | I went my self to the PO and handed it to the post office lady |
| 18:37.16 | r0m|u | I than continue to talk to her about my issues with sims |
| 18:37.32 | r0m|u | she explain to me what I said earlier. |
| 18:37.37 | p3nguin | I can't imagine someone would steal two random envelopes from you. |
| 18:38.06 | p3nguin | I'd guess the chances of even one being stolen would be low. |
| 18:38.32 | r0m|u | well I dont know... I am mad though. I hope that it makes it tomorrow |
| 18:38.47 | r0m|u | the other part is that this would be th 4th sim lost |
| 18:39.02 | p3nguin | Were the other two both to the same person? |
| 18:39.19 | r0m|u | No. two different people |
| 18:39.30 | p3nguin | The second try on each of them made it? |
| 18:39.37 | r0m|u | same post office |
| 18:39.47 | r0m|u | Yes they did. |
| 18:39.50 | p3nguin | Sounds like a conspiracy. |
| 18:40.23 | p3nguin | Somewhere there's a phone geek sitting there rolling in sims. |
| 18:40.30 | r0m|u | what ever it is is not fun. I loose money :( |
| 18:40.55 | r0m|u | well not on you since they are gifts but the other two where since they where sold threw gsm forums |
| 18:41.17 | p3nguin | I'd be pretty upset it it happened to me more than zero times. Really upset if it happened more than once. I don't know how I'd react if it happened four times. |
| 18:41.51 | p3nguin | They'd probably ask me to leave. |
| 18:42.14 | r0m|u | well I am hoping this is not the fourth time. Ill wait till tomorrow before I bring hell to them |
| 18:42.57 | p3nguin | I'll call the PO here and ask how long it should take. What city is it being mailed from? |
| 18:43.18 | r0m|u | Spring |
| 18:43.30 | r0m|u | actually Houston |
| 18:44.45 | p3nguin | Do you happen to know the last two digits on the zip code? |
| 18:44.50 | p3nguin | They have a bunch. |
| 18:44.58 | r0m|u | 05 |
| 18:45.11 | p3nguin | I'm going to call right now. |
| 18:45.25 | r0m|u | let me know what they say. |
| 18:46.18 | r0m|u | p3nguin, are this module safe to unload as per the guide? http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html |
| 18:46.52 | r0m|u | I am trying to strip shit down to a minimum.... Only what I need. |
| 18:47.05 | r0m|u | brb |
| 18:50.09 | p3nguin | She looked it up, and said for a first class letter if it was mailed today, it should be here on Monday, which would mean three days. She said it could take more or less, but that's the general estimate. |
| 18:50.30 | r0m|u | ok. |
| 18:50.56 | p3nguin | What module are you asking about? |
| 18:51.02 | r0m|u | well If it was mailed out on Tuesday since Monday there was no post office working than means it should be there tomorrow? |
| 18:51.36 | p3nguin | That's my calculation. |
| 18:52.00 | r0m|u | on the link I gave you the author explain how you should unload unneeded modules and proceeds to give you a list. |
| 18:52.11 | r0m|u | p3nguin, are this module safe to unload as per the guide? http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html |
| 18:52.20 | p3nguin | What module are you asking about? |
| 18:52.47 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 18:53.07 | r0m|u | p3nguin, http://pastebin.com/hid2Kv29 |
| 18:53.50 | p3nguin | I still can't answer your question accurately. |
| 18:54.04 | p3nguin | If you mean all the modules listed, my answer would be no it is not okay. |
| 18:54.06 | r0m|u | is it safe to use that list? |
| 18:54.15 | r0m|u | ok |
| 18:54.19 | r0m|u | I see. |
| 18:54.25 | p3nguin | If you want to know about a single module, I can probably determine if it's okay or not. |
| 18:54.54 | p3nguin | You have to determine if you need any of the things that you are trying to prevent from loading. |
| 18:55.08 | p3nguin | If you plan to use it, don't "noload" it. |
| 18:55.29 | p3nguin | If you don't want to use it, you probably don't need to load it, so noload would be okay. |
| 18:55.31 | r0m|u | Yeap thats what I have done so far. |
| 18:55.38 | r0m|u | here is what I have in my modules.conf |
| 18:55.40 | r0m|u | http://pastebin.com/DjraqkHC |
| 18:57.29 | p3nguin | That looks like a reasonable list of things to not load. |
| 18:57.38 | *** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net) |
| 18:57.39 | hardwire | moo |
| 18:58.40 | r0m|u | cool |
| 19:00.27 | p3nguin | In some systems, it is easier to figure out what modules you need and load only those. In other systems, it's easier to use autoload to load everything you have a conf for and noload only the ones you don't want. |
| 19:01.11 | r0m|u | I think in my case would be easire to just load what I need since is not a complex system |
| 19:06.00 | [TK]D-Fender | r0m|u, The real key items are DB and channel-drivers |
| 19:06.08 | [TK]D-Fender | r0Do you need IAX2? |
| 19:06.23 | [TK]D-Fender | If not that's one more thing.. |
| 19:06.42 | p3nguin | I guess I'm going to go to the butcher and get something for supper. There's really nothing else going on today, so I might as well. |
| 19:06.49 | r0m|u | [TK]D-Fender, Thanks for the pointer. |
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| 19:34.46 | Bob_Pierce | We have a customer who uses a Ground Start connection with their current PBX. We'd like to mock up some tests in our lab. Does anyone have any suggestions how to build and Asterisk Box with a Ground Start FXO port? What can I use for hardware for the FXO Port? |
| 19:35.58 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 19:36.28 | azv4 | would ride a bike on water for a copy of TD-500 Maintence Console 2.4 software |
| 19:38.26 | navaismo | Bob_Pierce TDM404EF |
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| 19:47.01 | arnotixe | hi all I have some issues ("not working :D") with IAX between two asterisk. "iax2 show peers" shows status OK at both servers. but all calls on the trunk are rejected =? verbose: http://pastebin.com/YzSVMH8D |
| 19:47.38 | [TK]D-Fender | pB your configs |
| 19:48.02 | Bob_Pierce | I have a TDM11B Would that work? |
| 19:49.52 | navaismo | with one fxo yes |
| 19:54.48 | arnotixe | ok this is config on both sides: http://pastebin.com/cyWYF756 |
| 19:56.21 | [TK]D-Fender | arnotixe, Your clien has no "secret", the other side does |
| 19:56.54 | arnotixe | ok in iax.conf on the client, definition section, secret is necessary? |
| 19:57.04 | arnotixe | even if it is registered above? |
| 19:57.09 | arnotixe | let me try |
| 19:58.12 | [TK]D-Fender | arnotixe, registration has nothing to do with authing calls |
| 19:58.42 | arnotixe | hm ok if I remember your hints from earlier today: registration is just saying to the other end: "I'm here" |
| 19:58.44 | arnotixe | ? |
| 20:00.16 | Bob_Pierce | navaismo thanks! I wasn't sure that would do ground start. |
| 20:01.08 | [TK]D-Fender | arnotixe, Funny thing ... I thought I was the one who said that... |
| 20:01.17 | arnotixe | yep |
| 20:01.17 | [TK]D-Fender | Bob_Pierce, I don't believe it does |
| 20:01.20 | arnotixe | <PROTECTED> |
| 20:01.42 | arnotixe | I just didn't get this: |
| 20:01.42 | arnotixe | [TK]D-Fender, Isn't the ":password" part of 'register => user:password@host' really redundant then? |
| 20:01.54 | arnotixe | I mean, the password could be read from iax.conf |
| 20:01.55 | [TK]D-Fender | arnotixe, No, it is separate. |
| 20:02.20 | timholum | Is it possible to set up chan_unistim to work with multiple ports? |
| 20:02.22 | arnotixe | ok I don't want to fight something that works anyway... |
| 20:02.32 | [TK]D-Fender | If you register it tells them the IP to call you at. What if someone hijacks your IP? |
| 20:02.33 | [TK]D-Fender | FRAUD |
| 20:03.01 | [TK]D-Fender | So no, it has no impact on the fact that all call attempts will be authed |
| 20:03.13 | [TK]D-Fender | unless you disable that... which is not smart |
| 20:03.29 | arnotixe | ah. I guess it's enabled by default? hehe |
| 20:03.34 | navaismo | Fender the ground start loop start or kewl start is relative to the configuration or hardware? |
| 20:03.49 | arnotixe | about fraud: some guy in africa loaded off a$1000 from me recently... |
| 20:03.56 | arnotixe | hijacking :( |
| 20:04.02 | [TK]D-Fender | timholum, You will have far more luck on the ML's for your request. We get about 3 people a year in here asking about UNISTIM. With you that only leaves 2. |
| 20:04.15 | arnotixe | but that wasn't asterisk's problem, it was one of the softphones |
| 20:04.26 | arnotixe | eh GSM/SIP |
| 20:09.15 | [TK]D-Fender | Bob_Pierce, Digium HW supports Loop & Kewl. Groundstat is only one direction IIRC. I'd call tech support before buying |
| 20:09.30 | [TK]D-Fender | Bob_Pierce, And their compatability chart doesn't cover this well. |
| 20:17.31 | Bob_Pierce | Thank You - It didn't seem clear from what I had read and tried if I should be able to make Groundstart work with a TDM11B |
| 20:19.40 | navaismo | maybe digium need to place a note in the DCAP manual where they explain the 3 signaling supported by asterisk |
| 20:20.45 | [TK]D-Fender | Not everything is in a manual. Welcome to open source :) |
| 20:23.35 | navaismo | i know but even in the manual of the card not say anything about that so I guess it support the 3 types of signalling |
| 20:24.33 | navaismo | As reseller we expect that kind of information. |
| 20:26.38 | [TK]D-Fender | "Get used to disappointment" - Westley (Cary Elwes), The Princess Bride (1987) |
| 20:26.42 | [TK]D-Fender | :) |
| 20:29.48 | [TK]D-Fender | checkout time, BBIAB |
| 20:31.00 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
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| 20:49.01 | EmleyMoor | Is there a third edition of the book? |
| 20:50.56 | Qwell | ~book |
| 20:50.57 | 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 |
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| 21:04.46 | pigpen | Hi all. I don't get in here often, but when I do, it is normally a good one. |
| 21:05.51 | pigpen | I am running 1.8.7.0, tried on 1.8.5 as well: when we enable call parking, the call is transferred into the lot and it recognized by it responding with the park number |
| 21:06.14 | pigpen | then it immediately reports that the caller got tired of waiting and the call is disconnected. |
| 21:06.27 | pigpen | The wait time is 45 seconds |
| 21:06.36 | pigpen | context is the same as the phones |
| 21:07.15 | pigpen | I brought this up the other day and p3nguin noted that I should upgrade from 1.8.5.0 to 1.8.7.0 to make sure it wasn't a known bug/fix. |
| 21:07.52 | pigpen | It was working on the same system on 1.8.4.0, however, we had sip segfaults on 1.8.4.x and moved to 1.8.5.0 for the fix. |
| 21:08.29 | pigpen | So I am just checking again to see if anybody has any great ideas. If so, I'll try them. If not, I'll file a bug. |
| 21:08.54 | _Corey_ | pigpen: I can confirm that 1.8.7.0 isn't doing that for me w/Park if that's of any help :( |
| 21:09.20 | pigpen | 32 or 64 bit? |
| 21:09.37 | *** part/#asterisk irroot (~irroot@41.51.73.114) |
| 21:09.40 | _Corey_ | 32 for me |
| 21:09.48 | pigpen | k. All 64bit here. |
| 21:09.57 | _Corey_ | are you explicitly setting 'parkingtime' in features.conf? |
| 21:10.09 | _Corey_ | (45 sounds like the default to me) |
| 21:10.24 | pigpen | you know?I'll make sure. I am am like 99% sure. |
| 21:10.33 | _Corey_ | Mine is set to 600sec |
| 21:10.42 | pigpen | because I like values defined, not "defaulted" |
| 21:11.08 | pigpen | yeah, defined. |
| 21:11.34 | _Corey_ | Have you experimented with that? |
| 21:11.43 | _Corey_ | (increasing/decreasing that is) |
| 21:11.45 | pigpen | This is also happening on several systems at several locations. Now, that being said, they are all imaged from the same image, all on the same hardware. |
| 21:12.15 | pigpen | no. I certainly can, but that would be really odd. But?I don't seem to get the easy problems. |
| 21:12.55 | pigpen | I have played with parkeddynamic yes/no/nonexistant |
| 21:12.58 | pigpen | but no dice. |
| 21:13.16 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 21:13.32 | _Corey_ | I'm not familiar with that code, so I don't know if it's going to yield much but you may want to crank up your debugging |
| 21:13.51 | pigpen | yeah, default is yes. |
| 21:14.08 | pigpen | and yeah, recursive is at like 12, but I may have to get some extras turned on. |
| 21:14.54 | *** join/#asterisk shido6_ (~shido6@nat/yahoo/x-iejpbqmacqtrwkul) |
| 21:15.12 | pigpen | I guess I can try moving it to a different parking "log" |
| 21:15.15 | pigpen | s/log/lot |
| 21:15.30 | pigpen | from 70 71-99 to 700 700 - 799 |
| 21:15.41 | pigpen | who knows, maybe I have something goofy in my dialplan. |
| 21:16.07 | pigpen | well, that being said, yes, I do. But that is just being creative. |
| 21:16.33 | _Corey_ | I'm using the standard 700 ad 701-720 |
| 21:17.34 | *** part/#asterisk joeflyde (~Gregory@63.239.202.68.cfl.res.rr.com) |
| 21:17.44 | pigpen | yeah, the gentoo builds are |
| 21:18.12 | _Corey_ | I'd try a stock features.conf from the samples under 1.8 to rule out anything "different" |
| 21:19.45 | pigpen | 70 - 71-99 |
| 21:21.50 | pigpen | funny enough, if you pickup the call real quick ( I mean real quick) you can grab it. |
| 21:22.03 | pigpen | but, yeah, changing it from 45 sec to 100 sec no change. |
| 21:22.28 | pigpen | another goofy thing is I see " == SIP/pstn-0000001a got tired of being parked" |
| 21:23.25 | pigpen | Could blind transfer cause an issue? |
| 21:23.29 | _Corey_ | Well, that's what you'd see when the timeout is reached under normal circumstances |
| 21:23.31 | pigpen | just thought about it. |
| 21:23.55 | pigpen | if it was transferred blind, it wouldn't know (or would it) where to call back to. |
| 21:24.54 | _Corey_ | no, I just tried a blind xfer |
| 21:26.49 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-woslijrjwxhkotna) |
| 21:27.33 | pigpen | well, I'll try changing it from my "default" of 70 to 700 and the park from 71 - 99 to 701 - 799 |
| 21:27.47 | pigpen | who knows. dumb luck can be a good thing sometimes. |
| 21:27.51 | pigpen | thanks for the help bty |
| 21:28.09 | _Corey_ | yah, no prob... sometimes it's easier to know what "works" elsewhere |
| 21:30.32 | pigpen | nope, didn't work. Right after it says the digits, it says "Spawn extension (dial-phones, s, 83) exited non-zero on 'Parked/SIP/pstn-00000005<ZOMBIE>'" |
| 21:30.36 | navaismo | pigpen Asterisk 1.8.7 here, I change the values for 70/ 71-99 and still works already one call in hold |
| 21:30.46 | pigpen | k. tks. |
| 21:30.48 | navaismo | testing with zoiper |
| 21:31.01 | pigpen | I remember seeing the ZOMBIE before now that I think of it. |
| 21:31.44 | pigpen | navaismo, are you 32bit or 64bit |
| 21:31.45 | pigpen | ? |
| 21:31.57 | navaismo | 64 bit Centos 6 |
| 21:32.04 | pigpen | k |
| 21:32.57 | pigpen | I can't imagine a dial plan feature that would cause this, as the parking feature is just that?a feature. |
| 21:33.02 | pigpen | no real setting for it in the dial plan. |
| 21:33.04 | pigpen | right? |
| 21:33.52 | navaismo | just include the parkedcalls context into your context phones or wahtever you called the context |
| 21:34.22 | pigpen | yeah, all is good there. |
| 21:34.40 | pigpen | that is mostly for where to exit to. |
| 21:35.34 | pigpen | Here is my box: |
| 21:35.35 | pigpen | Linux exivoice 2.6.38-hardened-atom-r2 #1 SMP Wed May 4 16:21:47 CDT 2011 x86_64 Intel(R) Atom(TM) CPU D525 @ 1.80GHz GenuineIntel GNU/Linux |
| 21:35.59 | pigpen | We have 10 deployed like this, will have another 380 or so to go. |
| 21:36.22 | navaismo | w |
| 21:36.24 | pigpen | the 64bit atoms have been pretty sweet. |
| 21:36.35 | pigpen | Running Gentoo. |
| 21:37.01 | pigpen | Business partner is a Gentoo Kernel dev (along with strongswan and others) |
| 21:37.16 | pigpen | but this doesn't seem like a system issue. |
| 21:37.24 | pigpen | So verdict? keep trying shit or bug? |
| 21:38.11 | pigpen | One last idea: I'll ditch the dial plan. Make it virgin. Just enough to get a call parked and unparked. |
| 21:38.30 | pigpen | if it happens with that, there is an issue. If not, I must be mucking it up with something. |
| 21:38.35 | pigpen | sound logical? |
| 21:48.49 | navaismo | no idea |
| 21:51.49 | pigpen | I am trying another approach at the moment. Calling phone to phone and trying the park. |
| 21:52.07 | pigpen | This way it keeps it from jumping context. |
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| 22:04.33 | hardwire | meh |
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| 22:32.22 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 22:32.40 | SeRi | p3nguin, you avail? |
| 22:33.22 | SeRi | does any body use any type of backup provider? |
| 22:33.29 | SeRi | for fail over that is.. |
| 22:34.42 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 22:34.59 | Dovid | hellp all |
| 22:35.17 | Dovid | hello* |
| 22:35.44 | hardwire | yellow! |
| 22:35.49 | Dovid | red !!! |
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| 23:02.34 | *** join/#asterisk navaismo (~ubuntu@fixed-203-96-202.iusacell.net) |
| 23:05.20 | citywok | i :heart: the idiots at bandwidth.com. 3 years of hell with them, we cancelled, they screwed it up and kept billing. also they kept giving us service for 5 weeks after our cutoff date (they refunded us the overbilling) |
| 23:05.43 | citywok | thanks for the 100,000 free minutes! lol |
| 23:08.01 | p3nguin | seri: Yes. |
| 23:08.47 | SeRi | p3nguin, what do you use? do you an sub account in .ms? |
| 23:08.54 | SeRi | use* |
| 23:10.30 | p3nguin | (1733.25) <SeRi> p3nguin, you avail? |
| 23:10.33 | p3nguin | (1808.00) <p3nguin> seri: Yes. |
| 23:10.59 | SeRi | oooo. |
| 23:11.17 | SeRi | p3nguin, do you use any type of fail over account? |
| 23:13.18 | citywok | SeRi: it's a good idea to have a second provider to make calls if your primary goes down. we use flowroute as our backup. |
| 23:14.52 | p3nguin | No. |
| 23:15.59 | SeRi | p3nguin, you never had any issue to have to resort to a backup? |
| 23:16.00 | p3nguin | I can't think of a time where voipms wasn't able to send my calls out. |
| 23:16.05 | SeRi | ah |
| 23:16.08 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:16.11 | SeRi | Thats what I was looking for. |
| 23:16.14 | SeRi | Thanks. |
| 23:16.37 | citywok | if you are using bandwidth.com then you should have a backup :P |
| 23:17.10 | p3nguin | Perhaps in extremely high call volume cases, you'd be more likely to see an outage. My usage is pretty low, so there could have been tiny outages in between my calls and I would never know it. |
| 23:18.39 | SeRi | Thanks for the info |
| 23:21.22 | p3nguin | Anyone familiar with handling DoS from an organization where the abuse contact for the org won't respond to voice messages left? |
| 23:21.56 | citywok | p3nguin: i had that happen, i got blasted 40 or 50gb worth of traffic trying to scan for a valid extension/password |
| 23:22.15 | citywok | after no response i ended up having the NOC at the datacenter block it for us |
| 23:22.21 | p3nguin | I should do more calling through flowroute and see if I like the service quality. I send only a low percentage of my calls through them, but those calls seem to be good. |
| 23:22.35 | citywok | i haven't had a single call quality issue with flowroute |
| 23:23.00 | citywok | i really hate their stupid software that tries to determine if you are making unusual calls and blocks you though (for international) |
| 23:23.13 | citywok | every time we start calling a new country we get blocked. lol. |
| 23:24.24 | p3nguin | This DoS is just a UDP flood with no distinct purpose other than to deny me of service. I've made several calls, left a couple messages. What's next? |
| 23:24.38 | citywok | do you have an ISP that will help? |
| 23:24.43 | p3nguin | Not mine. |
| 23:24.50 | citywok | our datacenter NOC was able to block traffic for us |
| 23:24.58 | p3nguin | I mean my ISP wouldn't do anything. |
| 23:25.03 | citywok | yea, i figured |
| 23:25.12 | p3nguin | They just null the IP address after a while and give me a new one. |
| 23:25.35 | citywok | the same person hammering you? is it so bad even iptables dropping them isn't enough? |
| 23:26.00 | p3nguin | It's one IP address, and it's pf rather than iptables. |
| 23:26.19 | p3nguin | Dropping the packets still uses my bandwidth and CPU. |
| 23:26.29 | p3nguin | (which I'm sure is their goal) |
| 23:26.46 | citywok | yea that's annoying as hell. digium's solution for the AA50 to this problem was to put a device in front of the AA50 and block it there. |
| 23:26.57 | p3nguin | Someone is pissed off that I won't give him root access to a server, so now I get a UDP flood. Today is the second time I noticed it. |
| 23:27.18 | p3nguin | I'm sure I know who it is. The timing is too coincidental to not be the person who I think it is. |
| 23:27.19 | citywok | on a bigger connection, flood him back. lol |
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| 23:27.59 | citywok | figure out who his host is and complain to his host directly (not necessarily the abuse on the whois) |
| 23:28.02 | p3nguin | It's coming from University of Toronto. |
| 23:28.18 | citywok | oooh universities are generally pretty good about handling complaints. interesting. |
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| 23:28.40 | citywok | i'd just call their student tech support line and raise hell :P |
| 23:28.51 | citywok | i'm sure they ahve a phone# for students/staff to call that can be found |
| 23:29.09 | citywok | (i used to work at the University of Washington's IT department) |
| 23:29.14 | p3nguin | If I can get cooperation from the Uni to find out if it is the person I think it is, I'll give the info to his boss and I'd imagine he'll be fired. |
| 23:29.37 | citywok | if you know what department they are in it's even better. lol. |
| 23:30.00 | p3nguin | The person I think is responsible for it does not work there. |
| 23:30.03 | citywok | but if you call and say John XXX has been flooding me with traffic burning up my bandwidth AND yours and complain enough they should get you the right person |
| 23:30.40 | p3nguin | I just need to build the case to find out if it can be traced back to him somehow. |
| 23:30.41 | citywok | is it a student or an employee? |
| 23:30.52 | citywok | if you have the ip it's ocming from that's all they need |
| 23:30.59 | citywok | they should know what port that IP is on and whose office/dorm that is. |
| 23:31.10 | p3nguin | No clue, the person I think is responsible has no relation to the Uni that I know of. |
| 23:31.11 | citywok | at the UW they would simply shut off the port until the person called in to say wtf |
| 23:31.24 | p3nguin | I have the IP address. I have gigs of firewall log to show. |
| 23:31.46 | citywok | brb5min |
| 23:31.55 | p3nguin | You think I should call again tonight or wait until business hours? |
| 23:32.19 | citywok | at the UW biz hours was the only time we were open. my guess is it's the same. |
| 23:32.31 | citywok | noc was 24x7 but finding their number may be more difficult |
| 23:33.10 | p3nguin | I haven't tried to find any other numbers yet. I was hoping the abuse contact would be good enough, but it hasn't so far. |
| 23:35.38 | p3nguin | It's still going on right now. |
| 23:36.06 | p3nguin | I've determined that it's better to just accept the useless packets and block any responses that my system may try to give back. |
| 23:37.24 | p3nguin | Odd. The source port is static, but my target port appears to be completely randomized. |
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| 23:42.36 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 23:43.18 | p3nguin | Blocking it seems to do more harm than good. |
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