| 00:02.53 | lauris | is it possible to extract disposition and billsec through agi on h extension? |
| 00:04.34 | p3nguin | Most likely. |
| 00:06.18 | lauris | do you know how? it's not included by default - http://pastebin.com/ar7qZTvb |
| 00:19.46 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 00:29.12 | SeRi | p3nguin: for sure th 501 is bad. |
| 00:46.45 | p3nguin | lauris: You'll have to write it into your script. |
| 00:46.54 | p3nguin | seri: How did you finally determine it is bad? |
| 00:55.44 | SeRi | p3nguin: I cant hear novody out of it at times |
| 00:55.51 | SeRi | nobody* |
| 00:57.13 | SeRi | p3nguin: my resource of 25 is now online and first one to ring |
| 00:57.16 | SeRi | next is my android |
| 00:57.32 | SeRi | :D |
| 00:57.45 | SeRi | I guess it need it time to show up as a resource |
| 00:58.09 | *** join/#asterisk coppice (~coppice@m121-202-50-109.smartone.com) |
| 01:04.23 | SeRi | p3nguin: how can I have my gv looup cnam? |
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| 01:47.05 | p3nguin | You'll have to look up the name externally, just like for any other type of phone service. |
| 01:47.25 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 01:48.06 | p3nguin | Dip into a cnam db and write it to the channel. |
| 01:50.36 | *** join/#asterisk coppice (~coppice@m121-202-50-109.smartone.com) |
| 01:53.05 | SeRi | p3nguin: ok |
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| 02:34.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 02:50.01 | k-man | I bought 9 x snom 300 phones second hand the other day. they were very cheap |
| 02:50.14 | k-man | are they OK? |
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| 05:02.18 | p3nguin | seri: What do you think about Toshiba and Fujitsu? |
| 05:44.14 | *** join/#asterisk LiuYan1 (~liu.yan@211.154.128.171) |
| 05:51.23 | WIMPy | k-man: If that's the same series as 320/360/370, then yes. |
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| 06:23.21 | k-man | WIMPy, yeah I believe it is the lowest spec model in that range |
| 06:23.39 | k-man | 2 line small LCD |
| 06:24.28 | WIMPy | At least it seems still supported. |
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| 07:10.32 | schmidts | good morning |
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| 08:19.32 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 08:19.44 | IsUp | hello |
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| 08:20.18 | WIMPy | IsUp GotUp |
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| 08:21.32 | IsUp | :) |
| 08:28.14 | ollii | ehlo |
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| 08:34.43 | IsUp | 503 bad sequence of commands. |
| 08:35.05 | ollii | 404 not found |
| 08:35.32 | IsUp | anyone using "nagios" or "monit" here? |
| 08:35.39 | ollii | yes |
| 08:36.05 | IsUp | which one? both? :p |
| 08:36.09 | ollii | first ne |
| 08:36.15 | ollii | s/ne/one/g |
| 08:36.50 | IsUp | are you monitoring asterisk (channels, peers etc) or connection? |
| 08:36.56 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 08:37.24 | ollii | yes |
| 08:37.54 | ollii | asterisk process, asterisk alive via AMI and some high priority peers like our media gateway |
| 08:38.45 | ollii | and sip stack |
| 08:38.57 | IsUp | is that possible to monitor "logs" with nagios? "grep 'UNREACHABLE' /var/log/asterisk/full" for example |
| 08:40.24 | ollii | you want to watch sip peers? |
| 08:41.14 | IsUp | ollii: i want to watch my net connection, and yes sip peers |
| 08:41.21 | ollii | http://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip/details |
| 08:41.25 | WIMPy | Logs? What about AMI? |
| 08:41.49 | IsUp | WIMPy: it was an example, i have some AGI verbose outputs in my logs, so i want to check them actually |
| 08:42.34 | WIMPy | If it's from AGI, why don't you actively trigger whatever you need? |
| 08:43.26 | IsUp | AGI is a remote host actually |
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| 09:07.03 | *** join/#asterisk Yourname`` (~Yourname@121.97.255.134) |
| 09:07.22 | Yourname`` | Hi. Does it matter if under peer disallow is top and allow is somewhere down below? |
| 09:07.37 | Yourname`` | Like disallow=all nat=yes qualify=yes allow=ulaw |
| 09:08.08 | WIMPy | Yes. It has to be in that order. |
| 09:13.24 | dijib | any late nighters in here? |
| 09:13.34 | dijib | nothing to report, reporting in |
| 09:16.44 | Yourname`` | WIMPy: You mean it doesn't matter if there's anything in between as long as disallow comes first? |
| 09:17.11 | WIMPy | correct |
| 09:18.57 | Yourname`` | Thanks! |
| 09:19.11 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:19.15 | Yourname`` | What about the deny and allow IP restrictions.. do they interfere with audio, by any chance? |
| 09:19.25 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 09:20.08 | WIMPy | That's only for SIP, *I think*. |
| 09:20.42 | Yourname`` | Cool, appreciate it. |
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| 09:29.34 | IsUp | Yourname``: it's deny= and permit=, not 'allow/disallow' |
| 09:31.01 | Yourname`` | That too IsUp :) |
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| 11:05.36 | *** join/#asterisk mac|gyver (~sanderrij@2001:960:66e:0:216:cbff:fead:1d5) |
| 11:07.06 | mac|gyver | I have a problem with incoming calls. It finds the peer, but results in 401.. username mismatch, have <trunkname>, digest has <anonymous> |
| 11:08.16 | IsUp | mac|gyver: pastebin your peer, mask passwords |
| 11:09.46 | mac|gyver | I will.. strange thing is it uses the outgoing trunk name, not incoming |
| 11:10.25 | mac|gyver | http://pastebin.com/u1Vr2TjT |
| 11:11.40 | IsUp | mac|gyver: which host sending you calls? because your host is not matching as my opinion |
| 11:11.48 | mac|gyver | sip.xs4all.nl |
| 11:12.10 | IsUp | mac|gyver: okay just try "host=sip.xs4all.nl", dont use &, reload and place a call for test |
| 11:12.39 | mac|gyver | that won't work, because sometimes the provider also uses the other IP. I can skip dynamic though |
| 11:15.04 | IsUp | mac|gyver: i am not so sure if you can use a dynamic ip on peer. |
| 11:15.15 | mac|gyver | it's gone.. |
| 11:15.31 | IsUp | mac|gyver: try template, define 3 other hosts. |
| 11:15.38 | mac|gyver | hm? |
| 11:17.12 | mac|gyver | hmm removing the &<ip> fixes it |
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| 11:17.32 | mac|gyver | until they start using the other IP, then it'll break again |
| 11:18.04 | IsUp | mac|gyver: then register to them, they should be using static ip. |
| 11:18.33 | IsUp | mac|gyver: if you know all IPs that calls coming from, define them one by one, but use template. so you dont have to type context codecs and other settings again. |
| 11:19.08 | mac|gyver | ah clear.. I'll try this for now, and use template when needed.. |
| 11:19.12 | mac|gyver | thanks |
| 11:19.40 | IsUp | mac|gyver: np |
| 11:20.56 | mac|gyver | about complaining to the provider. I found threads dating back to 2005 complaining about the same thing. So I think we're considering another provider :-) |
| 11:21.27 | IsUp | :) |
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| 12:17.58 | nny | hmmph |
| 12:18.43 | nny | : No closing parenthesis found? 'SipAddHeader(P-Asserted-Identity: <sip:6782185796@arson.com' at line 175 of extensions.conf FROM exten => _1XXXXXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${6782185796@arson.com};otg=sc6960>) |
| 12:18.56 | nny | <PROTECTED> |
| 12:19.01 | nny | this is suggested by a voip vendor |
| 12:19.07 | nny | the otg is breaking it |
| 12:19.26 | IsUp | }\;otg |
| 12:19.26 | nny | but not sure what/where/why they are making it so complicated. Any help in correcting their statement appreciated |
| 12:19.32 | nny | IsUp: heh |
| 12:19.45 | nny | IsUp: we meet again, and um.. same thing as before with them. I'll try it |
| 12:20.09 | nny | the last suggestion worked great, seems NO special chars in register strings though they sent me a ? after the first one |
| 12:21.24 | nny | SipAddHeader(P-Asserted-Identity: <sip:${6782185796@arson.com}\;otg=sc6960>) |
| 12:21.27 | nny | ? |
| 12:21.52 | nny | sorry changed it back to having the ${}. not sure if that's needed or not for this |
| 12:22.03 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:22.30 | IsUp | exten => _1XXXXXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:6782185796@arson.com\;otg=sc6960>) |
| 12:22.39 | nny | IsUp: will try thanks |
| 12:23.09 | *** join/#asterisk _omer (~omer@119.152.113.153) |
| 12:23.38 | _omer | hello |
| 12:23.39 | nny | IsUp: no error, thanks! |
| 12:23.53 | nny | IsUp: heh they're suggested stuff has got me all discombobulated with proper syntax |
| 12:23.55 | _omer | I have a queuemetrics related question, Anyone have experience with QueueMetrics ? |
| 12:23.57 | nny | their* |
| 12:24.06 | IsUp | nny: :) |
| 12:24.24 | nny | IsUp: apparently I am helping a voip vendor learn how to setup trunks by proxy.. :\ |
| 12:24.42 | IsUp | brb |
| 12:27.37 | _omer | QueueMetrics QM.Realtime returns Array header but not any data.........anyhelp ??? |
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| 12:31.58 | nny | lol AAaaand back to the vendor |
| 12:32.27 | nny | <PROTECTED> |
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| 12:33.07 | nny | this has to be the most involved setup yet for this.. probably some horrific media gateway on their end |
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| 13:29.03 | Yourname` | Hi. I have an asterisk server, eyebeam softphones on windows 7, and a gsm gateway all behind NAT. The asterisk server has the gsm gw as it's sip trunk and 2 dahdi lines coming in. Ports have been forwarded to the pbx IP. Windows firewalls are disabled. I'm having one way audio issues. Sometimes complete dead air. Asterisk 1.8.7.0. SIP trace: http://pastebin.com/Tm5HfgTB |
| 13:29.43 | Yourname` | [ADDTIONAL INFO] Hi. I have an asterisk server, eyebeam softphones on windows 7, and a gsm gateway all behind NAT. The asterisk server has the gsm gw as it's sip trunk and 2 dahdi lines coming in. Ports have been forwarded to the pbx IP. Windows firewalls are disabled. I'm having one way audio issues. Sometimes complete dead air. *43 shows a little bit of choppiness and also makes it sound |
| 13:29.44 | Yourname` | like I'm farrr from the MIC. Asterisk 1.8.7.0. SIP trace: http://pastebin.com/Tm5HfgTB |
| 13:32.38 | Yourname` | Oh, also, when the call comes in through DAHDI, 9 out of 10 times there's full on dead air on both sides. |
| 13:35.17 | IsUp | ~NAT |
| 13:35.17 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 13:35.44 | *** join/#asterisk Quintana (~sylvain@91.194.178.68) |
| 13:36.15 | Yourname` | IsUp: All set and done too. Sorry I didn't mention that in the original long post. |
| 13:38.16 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:51.23 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 13:52.26 | asteriskATmarmuD | hi guys. I need a complete list of all asterisk 1.4 hangupcauses. all I found is the one from voip-info which does not seem to be complete: http://www.voip-info.org/wiki/view/Asterisk+cmd+Hangup |
| 13:54.43 | WIMPy | asteriskATmarmuD: Q.850 |
| 13:55.14 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:55.57 | asteriskATmarmuD | WIMPy: thx a lot! |
| 13:56.50 | Yourname` | Nobody? |
| 14:02.53 | asteriskATmarmuD | WIMPy: I gotta ask and assure myself, is that all related to asterisk 1.4? |
| 14:04.31 | WIMPy | No, it's not related to Asterisk at all. |
| 14:05.14 | *** join/#asterisk serafie (~erin@nat/digium/x-mzejcsytzncaqcom) |
| 14:07.26 | asteriskATmarmuD | WIMPy: ok, but asterisk uses the same defined codes? |
| 14:08.00 | WIMPy | It only gives you what it received from the network. |
| 14:10.19 | TheCops | Looks like its gonna be a christmast without snow |
| 14:10.30 | TheCops | oops sorry |
| 14:10.50 | WIMPy | Did you lock it up? |
| 14:14.13 | asteriskATmarmuD | I am looking at the codes right now. hmm, my network is an SIP-TO-ISDN gateway... but we won't be able to use more that the ones stated in that document? |
| 14:14.50 | asteriskATmarmuD | I mean these codes are all we can use? I can't any "redirection" code :( |
| 14:15.06 | leifmadsen | asteriskATmarmuD: look in the code -- <source>/include/asterisk/causes.h |
| 14:15.14 | WIMPy | In that case I wouldn;t exprct HANGUPCAUSE to contain sensible values. |
| 14:15.31 | Yourname` | Anyone around for helping me out on audio issues? |
| 14:15.40 | WIMPy | In that case I wouldn't expect HANGUPCAUSE to contain sensible values. |
| 14:16.09 | asteriskATmarmuD | thx guys. will go on on my own. your help is really appreciated. |
| 14:16.17 | WIMPy | The translations from SIP are few and last time I looked at them partially questionable. |
| 14:17.00 | WIMPy | Maybe your GW sends some SIP header containing the original value. Take a look with sip debug. |
| 14:17.36 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 14:17.51 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:18.05 | asteriskATmarmuD | WIMPy: we can even tell our device to map certain ISDN codes to defined SIP-codes. these should printed using "sip debug" |
| 14:18.30 | asteriskATmarmuD | WIMPy: but asterisk 1.4 can't really use SIP codes (at least in the dialplan) |
| 14:19.28 | WIMPy | Yes, I think that only works for INVITEs. |
| 14:20.48 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kgepcrkuvraiffxk) |
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| 14:22.11 | kaldemar | asteriskATmarmuD: you use numbers for cause codes that are mapped to SIP ones. |
| 14:23.10 | asteriskATmarmuD | kaldemar: is that a question or did you just tell me how asterisk works? |
| 14:24.06 | kaldemar | asteriskATmarmuD: not a question. see hangup_cause2sip in chan_sip.c for mappings. |
| 14:24.24 | asteriskATmarmuD | kaldemar: great many thanks |
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| 14:25.07 | kaldemar | all numbers are not there in the comments, you'll need to have a peek at include/asterisk/causes.h as well. |
| 14:25.36 | leifmadsen | heh, I mentioned that 10 mins ago as well :D |
| 14:27.08 | asteriskATmarmuD | kaldemar: causes.h already opened. you are so good to me :) |
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| 14:28.37 | CGMChris | It's too early in the morning for header files. |
| 14:28.47 | TheCops | lol |
| 14:29.07 | TheCops | its not that hard :p |
| 14:29.41 | CGMChris | You're right...the header files are the easy part... its the c file that can get hairy. |
| 14:31.10 | n1x0n | Hello, I have register foo:pass@egg/100 , then 100 does exten 100,1,Dial(SIP/foo1&SIP/foo2...,30,r) so that all phones ring when I get incomming calls - it all works fine excluding the caller id, do I need to do some "magic" (like storing it in a temp variable etc..) to get callerid to be passed further down ? |
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| 14:58.12 | leifmadsen | n1x0n: no, although possible you wan the 'o' option to dial -- or if this is SIP (which it looks like) you may need trustrpid=yes and sendrpid=yes in sip.conf |
| 14:59.14 | n1x0n | leifmadsen: thank you very much , I'll try your suggestions. |
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| 15:52.59 | *** join/#asterisk kl4m (~klamontag@gw2.noc1.sys-tech.net) |
| 15:58.25 | kl4m | Hi, I'm looking for a way to make voicemail transfers (option 8 on a message) work where my mailboxes have a prefix (ex.: clientname101) . Transfers currently give me e.g. " forward_message: '101' is not a valid mailbox". Voicemail is working fine otherwise. I'm using realtime voicemail with MySQL. Any ideas? |
| 15:59.01 | *** join/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 15:59.05 | DagMoller | hi |
| 15:59.19 | DagMoller | app_fax or res_fax? |
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| 16:09.07 | mjordan | kl4m: app_voicemail's menus are compiled into the application. As such, their behavior is what it is - in the case of forwarding a message, it either uses the output of app_directory or it collects DTMF digits and uses that to look up a mailbox. There isn't a way of prepending anything to that input. |
| 16:10.22 | kl4m | OK I'll patch something |
| 16:24.22 | *** join/#asterisk ven0m (~ven@87.69.145.169) |
| 16:24.52 | ven0m | Hi. I've connected sip trunk to my server. For reason, when i dial, i get Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21 |
| 16:24.56 | ven0m | what could be the problem? |
| 16:25.09 | ven0m | I also got Dial failed due to trunk reporting BUSY - giving up |
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| 17:38.16 | leifmadsen | ven0m: usually means the other side rejected the call -- look at the SIP trace and see what the problem is |
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| 17:49.24 | r0m|u | any ideas what this means? res_jabber.c:2060 aji_client_info_handler: User email@gmail.com/TalkGadgetF1820CF3 does not support discovery. |
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| 17:53.22 | *** join/#asterisk jimbo_uk (~IceChat77@84.12.253.146) |
| 17:53.35 | jimbo_uk | Hi all |
| 17:53.42 | jimbo_uk | Anyone in UK out there? |
| 17:53.57 | Qwell | ~ask |
| 17:53.57 | 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. |
| 17:54.00 | paulc | I'm from the UK, but now in Canada |
| 17:54.21 | jimbo_uk | Hi Paul, |
| 17:54.34 | jimbo_uk | nice, canada is lovely! I'm touting for a UK'er though! |
| 17:55.08 | paulc | Canada's alright, although the weather's a bit mardy today: www.katkam.ca |
| 17:55.12 | paulc | what you after a UK'er for? |
| 17:55.22 | jimbo_uk | looking for some long term help |
| 17:55.45 | Qwell | jimbo_uk: You aught to try the asterisk-biz mailing list. |
| 17:56.06 | jimbo_uk | sounds good |
| 17:56.22 | Qwell | be specific. Don't just say "I need someone in the UK for something." |
| 17:56.30 | *** join/#asterisk tyman (~tyman@216.64.184.67) |
| 17:56.35 | jimbo_uk | lol |
| 17:56.41 | jimbo_uk | noone on -biz but me!!! |
| 17:56.51 | jimbo_uk | ok, i'll be specific |
| 17:56.58 | Qwell | That would be why I said mailing list... |
| 17:57.02 | jimbo_uk | we're recruiting for asterisk experts |
| 17:57.09 | jimbo_uk | ah, ok |
| 17:58.08 | paulc | jimbo_uk: Have passport, will travel? ;-) That said, I'm pretty settled in Canada.. but I was talking to another guy from back home last week who was looking for people, kicking off a biiig project. Related? |
| 17:58.35 | jimbo_uk | possibly, don't think so though |
| 17:58.44 | jimbo_uk | we always have big projects on the go. |
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| 18:09.03 | jimbo_uk | Hey Paul, |
| 18:09.14 | jimbo_uk | We're looking for a full time role.... |
| 18:09.23 | jimbo_uk | i guess it's not up your street? |
| 18:15.21 | *** join/#asterisk libryder (~david@209.33.214.243) |
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| 18:17.56 | libryder | i'm using mixmonitor currently so i can move the recorded file after mixmonitor is finished... is there a way to do this with monitor? |
| 18:19.59 | libryder | orrr is there a way to get mixmonitor to separate audio from the two channels? |
| 18:20.27 | p3nguin | Monitor() records the channels separately. |
| 18:21.04 | libryder | i'm using mixmonitor right now because i can pass it a command to move the recorded file to an NFS mount |
| 18:21.06 | p3nguin | MixMonitor(), as the name should imply, records the channels mixed. |
| 18:21.26 | libryder | but i'd like to have separate files for each part of the stream |
| 18:21.33 | libryder | both channels* |
| 18:22.41 | p3nguin | I don't know what to tell you -- MixMonitor mixes as it records. |
| 18:23.23 | cmendes0101 | If you want to use Monitor your going to have to do System or something after to execute the move |
| 18:23.40 | cmendes0101 | I dont believe monitor has a exec feature like mixmonitor |
| 18:24.07 | paulc | jimbo_uk: delayed reply, sorry... uh.. "could be" - always open to talking but kinda half settled here.. talk is free though :-) |
| 18:24.38 | libryder | cmendes0101: thanks |
| 18:24.50 | libryder | p3nguin: the answer was pretty simple, no? |
| 18:25.04 | paulc | libryder: you can use Monitor then call a macro afterwards I think - let me check something here.. |
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| 18:25.28 | p3nguin | Yes, the answer WAS simple. |
| 18:25.42 | cmendes0101 | yah might want to do a macro since it would be 2 files to clean the context |
| 18:25.55 | paulc | libryder: yes - Set MONITOR_EXEC to whatever you want to run after Monitor is complete. It passes in 3 arguments I think.. left, right, and mix filenames |
| 18:25.56 | r0m|u | p3nguin: p3nguin whats going on. |
| 18:26.25 | cmendes0101 | doesnt MONITOR_EXEC just replace teh application it uses to mix? default:sox |
| 18:26.35 | cmendes0101 | oh wait wrong thing |
| 18:26.42 | paulc | libryder: in mine, I call a PHP script that does some sox and lame stuff, mixing the channels into 1 file (caller on the left, agent on the right) then converting to MP3 and moving from ram disk to NFS |
| 18:26.43 | cmendes0101 | nvm |
| 18:28.40 | libryder | paulc: so you basically have an external script that does all the work for monitor? |
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| 18:31.19 | paulc | libryder: yeah - I have this in my dialplan: same => n,Set(MONITOR_EXEC=/var/www/..../ws/CLI_ProcessRecording_OldSox.php) |
| 18:31.53 | libryder | and monitor is called after that? |
| 18:32.18 | paulc | libryder: then that PHP script has a #!/usr/bin/php at the begining and then does a bunch of stuff with the database (based on filename passed in - it's related to the internal call ID), and it shells out to call sox and lame to do mixing and converting |
| 18:32.25 | paulc | yeah - Monitor is the very next line in the dialplan |
| 18:32.46 | paulc | I can give you the PHP script if you like (I've got 2 versions - different distros used different versions of sox) |
| 18:33.26 | libryder | yeah that would be awesome |
| 18:33.46 | paulc | libryder: PM me your email |
| 18:33.50 | cmendes0101 | you need to have the seperate channels plus a mixed version? |
| 18:35.01 | libryder | cmendes0101: yeah one side of the call will be sent to a speech analytics system and the mixed version goes to a web gui where people can listen to their calls |
| 18:36.04 | cmendes0101 | ah ok |
| 18:39.41 | paulc | What speech analytics are you using? We use Nexidia (but not with Asterisk) |
| 18:41.37 | *** join/#asterisk bmg505 (~leon@196-209-101-126.dynamic.isadsl.co.za) |
| 18:42.06 | libryder | we're going to be using callminer - mostly for pci compliance |
| 18:45.20 | paulc | gotcha.. haven't seen their stuff myself.. Nexidia's stuff isn't bad for call center analysis.. we also got some bespoke work done for another app internally |
| 18:45.27 | paulc | PCI.. *shudder* hehe |
| 18:49.26 | p3nguin | r0m|u: I'm still trying to find a solution to this bug: https://bugzilla.novell.com/show_bug.cgi?id=733080 |
| 18:50.32 | r0m|u | p3nguin: ah! |
| 18:50.52 | r0m|u | I am still dealing with the choppiness issues. |
| 18:51.03 | r0m|u | looks like the issue is in the route as we talked before |
| 18:51.37 | p3nguin | My dhcpd uses Classless-Static-Route Option 121 or Classless-Static-Route-Microsoft Option 249, according to the client requesting the lease. |
| 18:51.37 | r0m|u | all calls from and to voip.ms using my works wifi with a sip client to voip.ne are perfect |
| 18:52.00 | r0m|u | the route shows no loss from wrok to voip.ms |
| 18:52.01 | carrar | p3nguin, Did you download the latest version of DHCPD from ISC? |
| 18:52.11 | r0m|u | so looks like comcast is defently at fault |
| 18:52.15 | p3nguin | No, because dhcpd isn't the problem. |
| 18:52.33 | p3nguin | dhcpd works correctly, giving both the static route and the default gateway. |
| 18:54.06 | p3nguin | If the dhcpd wasn't including both the static route and the default gateway, then I would blame the dhcpd. |
| 18:54.08 | carrar | Oh you are running dhcp server on a windows server anyways |
| 18:54.15 | p3nguin | No, I'm not. |
| 18:54.42 | carrar | try a different OS? |
| 18:56.09 | p3nguin | Yes. Windows XP clients accept the classless static route and the default gateway correctly. |
| 18:56.44 | *** join/#asterisk Beetlejooz (~IceChat77@12.237.61.93) |
| 18:57.03 | Beetlejooz | Hola |
| 18:57.07 | p3nguin | I'll look into upgrading dhcpd just to eliminate that as a variable. |
| 18:57.13 | carrar | Using latest ISC DHCP client? |
| 18:57.52 | carrar | I don't use opensuse |
| 18:58.07 | p3nguin | I don't use opensuse, either. |
| 18:58.24 | carrar | Should use a router to route networks :) |
| 18:58.26 | Beetlejooz | Can anyone point me toward a reliable SIP provider -other than vitelity- who supports call center traffic? Only doing B2B manual dials. |
| 18:58.27 | carrar | not PC's |
| 18:58.42 | p3nguin | I _AM_ using a router to route. wtf |
| 18:58.54 | carrar | Why are you adding static routes then? |
| 19:00.09 | p3nguin | Because there are two networks, each with a different router, but there is only one dhcp server. Clients from network A should have a static route to network B through router B. |
| 19:00.57 | carrar | The two routers should talk to each other |
| 19:01.00 | Beetlejooz | 2 networks should require only one router. |
| 19:01.02 | WIMPy | Isn;t that their default GW then? |
| 19:01.02 | carrar | have have the routes ont hem |
| 19:01.10 | carrar | let the routers figure out how to get to the other network |
| 19:01.16 | carrar | let the clients just haev a default |
| 19:01.42 | carrar | You ip helper addresses on the routers to forward DHCP requests |
| 19:01.44 | carrar | use |
| 19:02.12 | p3nguin | I guess that's one alternative approach, but not the one I was trying to use. Classless static routes are perfectly acceptable for this type of network... if the clients were not broken. |
| 19:02.14 | carrar | unless these are crqp routers and don't offer that |
| 19:02.27 | Beetlejooz | Are these cisco routers? |
| 19:02.31 | p3nguin | No. |
| 19:02.46 | p3nguin | And the routers aren't the problem. |
| 19:02.52 | carrar | they are the solution |
| 19:03.27 | p3nguin | The problem is that dhcpcd doesn't configure the default gateway route when Classless-Static-Route Option 121 was also received. |
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| 19:03.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:04.36 | p3nguin | I didn't notice the dhcpcd version mentioned in that guy's bug report. |
| 19:04.58 | Beetlejooz | use ip helper addresses like carrar said |
| 19:05.05 | Beetlejooz | save yourself the headache |
| 19:05.20 | p3nguin | CLASSID='dhcpcd 3.2.3'? Can't be version 3.2.3 ... I'm using version 5.x.x. |
| 19:05.24 | carrar | You'll find other not so popular dhcp options don't always work |
| 19:05.47 | carrar | or buggy |
| 19:06.19 | p3nguin | I don't need helper addresses -- there is only a single dhcp server. |
| 19:06.29 | carrar | which is why you DO need it |
| 19:06.32 | Beetlejooz | lol |
| 19:07.31 | Beetlejooz | how is the network with dhcp server going to get the bootp traffic if it doesn't know about it? |
| 19:07.38 | p3nguin | But adding a route to router A so that clients A can have a route to network B is something I will do since classless static routing is busted in dhcpcd. |
| 19:07.45 | WIMPy | I don't think I get the picture. Do you have two routers on each net? One for default and one for the other LAN? |
| 19:08.55 | p3nguin | There is one physical network. There are two subnets. There are two routers -- one internet gateway for each subnet. There is one dhcpd. |
| 19:09.01 | Beetlejooz | remember, routers don't forward broadcast traffic unless you tell them to |
| 19:11.11 | carrar | These default routers can't have two networks configred on them? |
| 19:11.19 | carrar | are they "DSL" modems? |
| 19:11.26 | p3nguin | Router B is a part of both subnets. |
| 19:11.35 | WIMPy | Very interesting setup, but I fail to see why you need an additional route. You need a node with IPs from bith subnets to route, that should probably be the router that alredy is the default anyway. |
| 19:12.16 | WIMPy | And router A is not on B? |
| 19:12.22 | p3nguin | correct |
| 19:12.40 | WIMPy | Any reason you can't change that? |
| 19:12.46 | Beetlejooz | You could always add a dhcp server to your other network and end the dilemma |
| 19:13.06 | p3nguin | What is that going to solve? The problem is not the dhcp server. |
| 19:13.13 | p3nguin | Clients in both networks get dhcp just fine. |
| 19:13.18 | WIMPy | How would that make a difference? |
| 19:14.11 | p3nguin | The problem, as I've said at least three times, is that dhcpcd does not configure a default gateway on the client when Classless-Static-Route Option 121 is also received. |
| 19:14.44 | WIMPy | Or the problem is that Router A is not on net B. |
| 19:15.08 | p3nguin | The dhcp server is giving out both Classless-Static-Route Option 121 *and* Default-Gateway Option 3 as it should be. |
| 19:15.18 | p3nguin | The problem, as I've said at least four times, is that dhcpcd does not configure a default gateway on the client when Classless-Static-Route Option 121 is also received. |
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| 19:16.14 | WIMPy | Why is Router A not in net B? |
| 19:16.22 | p3nguin | A Windows XP client sets both the static route and the default gateway, as it should, and as I expected dhcpcd on the Linux client to do. |
| 19:16.32 | carrar | p3nguin: http://www.osburn.com/p3nguin.jpg |
| 19:16.35 | WIMPy | And have you tried replacing dhcpcd? |
| 19:16.40 | carrar | is that correct per your design? |
| 19:16.53 | p3nguin | I'm not sure why... but I'm about to create a route on A to network B to work AROUND the broken dhcpcd on the clients. |
| 19:17.31 | WIMPy | That sounds like a sensible idea. |
| 19:18.09 | WIMPy | BTW: I don't know if it's better, but I always preferred dhclient over dhcpcd. |
| 19:18.16 | p3nguin | carrar: More like this: http://imagebin.org/index.php?mode=image&id=188014 |
| 19:18.42 | carrar | This is a logical view |
| 19:18.43 | Qwell | p3nguin: You have to specify the default gateway in the 121 |
| 19:18.48 | carrar | so same thing |
| 19:18.52 | WIMPy | dhcpcd does funny things like configure interfaces that aren't connected. |
| 19:19.00 | carrar | switch can be VLAN'd |
| 19:19.02 | Qwell | dhcp shouldn't be sending 121 and 3 |
| 19:19.05 | carrar | to split out the networks |
| 19:19.17 | p3nguin | uh, what? Shouldn't send 121 and 3? |
| 19:19.22 | carrar | or are you running two networks on the same single logical network? |
| 19:19.23 | Qwell | p3nguin: http://tools.ietf.org/html/rfc3442, DHCP Server Administrator Responsibilities |
| 19:22.01 | p3nguin | It says it should send both 121 and 3. But 121 should contain the static route and the default gateway, and 3 will contain only the default gateway (as it typically does). |
| 19:22.23 | *** join/#asterisk ruied (~ruied@po-217-129-155-146.netvisao.pt) |
| 19:22.36 | p3nguin | I tried to add the default gateway into the classless static route option, but it didn't let me have two routes. |
| 19:22.55 | p3nguin | That may be a limitation of the Vyatta software, or maybe I need to try harder. |
| 19:26.30 | p3nguin | When a Windows client asks for a lease, the dhcp server gives it Classless-Static-Route-Microsoft Option 249 and Default-Gateway Option 3, and the client configures correctly and as expected. |
| 19:26.52 | carrar | here you go |
| 19:26.52 | carrar | http://www.osburn.com/p3nguin.jpg |
| 19:26.56 | carrar | get the right router |
| 19:27.16 | p3nguin | I have the right routerS. |
| 19:27.27 | carrar | I would disagree |
| 19:27.45 | p3nguin | Of course you would. |
| 19:27.48 | carrar | heh |
| 19:28.30 | WIMPy | If just both of them knew about the other network, that would be good enough. |
| 19:28.35 | p3nguin | At some point, router A is going to be removed, and router B will deal with it all. |
| 19:28.50 | WIMPy | But only one might be nicer. |
| 19:28.57 | p3nguin | But for the time being, I just want it to work as it should. |
| 19:30.52 | WIMPy | That is you'd like dhcpcd to do what you like, not the setup to work? |
| 19:33.35 | r0m|u | p3nguin: I changed servers on voip.ms and now all my calls are going to misc_calls |
| 19:33.37 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 19:33.54 | p3nguin | If the correct operation of dhcpcd is to ignore Default-Gateway Option 3 when it also receives Classless-Static-Route Option 121, then I don't like the correct operation of dhcpcd. |
| 19:34.15 | p3nguin | r0m|u: Change the host in the peer and in the register statement. |
| 19:34.58 | r0m|u | ops for got the peer |
| 19:36.17 | r0m|u | p3nguin: I moved to chicago and calls now sound ok. mtr shows a clean path! |
| 19:36.23 | p3nguin | Good. |
| 19:36.35 | p3nguin | Atlanta might be closer, though. |
| 19:36.44 | r0m|u | Mhhhh true |
| 19:37.06 | p3nguin | Don't forget to change your DIDs to route to the correct server. |
| 19:37.12 | r0m|u | I did |
| 19:38.43 | *** join/#asterisk blizzow1 (~jburns@67.50.165.58) |
| 19:39.41 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 19:39.45 | p3nguin | The only real difference using Router A to route to Router B rather than clients having a static route to Router B is the extra hop when trying to reach Network B. |
| 19:39.56 | p3nguin | This isn't a problem, but a fact. |
| 19:40.16 | carrar | Router A may send a ICMP redirect to the client |
| 19:40.25 | carrar | so that the client sends it's traffic to router B |
| 19:40.59 | p3nguin | traceroute shows router A in the path between client A (on net A) and client B (on net B). |
| 19:41.21 | p3nguin | 1. Router A 2. Router B 3. Client B |
| 19:41.22 | carrar | thats because router A is the default |
| 19:41.24 | p3nguin | right |
| 19:41.43 | *** join/#asterisk vassilux (~vassilux@LPuteaux-156-14-48-156.w82-127.abo.wanadoo.fr) |
| 19:41.59 | p3nguin | With a classless static route, there are only two hops. 1. Router B 2. Client B |
| 19:42.16 | p3nguin | Both are acceptable, since it is a LAN. |
| 19:43.29 | *** join/#asterisk j0 (~dan@S01060017c562385b.va.shawcable.net) |
| 19:43.52 | p3nguin | <p3nguin> This isn't a problem, but a fact. <-------- this |
| 19:53.40 | carrar | problems don't exist, only opportunities |
| 19:53.44 | carrar | heh |
| 19:55.15 | carrar | Our Congress and Senate are missing their opportunity |
| 19:56.14 | vassilux | hi, I try to use Asterisk-GUI with TDMOE without success, so may someone did it ? I can add this function in Asterisk-GUI but I don't know where I can be start. |
| 19:56.52 | carrar | Thats a great opportunity for #asterisk-gui to answer your question. |
| 19:57.19 | vassilux | Yes but I asked without any response :-) |
| 19:57.32 | [TK]D-Fender | Join now! You can double their non-employee population! |
| 19:57.41 | r0m|u | rofl! |
| 19:59.57 | carrar | Hi TK!! |
| 20:01.30 | carrar | Lunching.....Gnocchi |
| 20:01.36 | carrar | err Lunch Time |
| 20:18.20 | *** join/#asterisk twanny796 (~twanny@78.133.49.164) |
| 20:20.25 | Beetlejooz[A] | is now away - Reason : bbiaf |
| 20:20.59 | Roadblock_RVA | Question about dial 0 to return to operator: If you are listening to someones voicemail message and you press 0 it should stop playing the voicemail and transfer the call? |
| 20:24.23 | [TK]D-Fender | Roadblock_RVA: It should leave app_voicemail and go to "o" if you have it defined in the current context. |
| 20:24.31 | pabelanger | Beetlejooz[A]: disable your auto-away notifications |
| 20:26.35 | Roadblock_RVA | That is what I had thought. It was setup by default in a PIAF install, and I'm assuming asterisk in general. |
| 20:26.46 | Roadblock_RVA | It however doesn't seem to be working. |
| 20:28.10 | [TK]D-Fender | better go check with them |
| 20:28.58 | Roadblock_RVA | I mostly wanted to try and make sure I wasn't barking up the wrong tree on how it was suppsed to be working. Thanks for the info. |
| 20:32.56 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:34.38 | *** join/#asterisk Ta^3 (~tacvbo@fixed-203-152-144.iusacell.net) |
| 20:36.20 | p3nguin | r0m|u: If you changed your route to use another voipms location, and that works well, then Comcrap actually isn't the problem? |
| 20:37.45 | r0m|u | p3nguin: Is taking a different route |
| 20:37.55 | r0m|u | it's not the same route to atlanta |
| 20:38.09 | p3nguin | Yeah, that's the point of a different pop. |
| 20:38.11 | r0m|u | the route to houston.voip.ms uses all comcast hops |
| 20:38.23 | r0m|u | all of the comcast hops have issues |
| 20:38.39 | r0m|u | so looks like a routing issue on comcast |
| 20:38.57 | *** join/#asterisk pdtpatrick1 (~ptaylor@12.249.4.226) |
| 20:39.05 | r0m|u | This is affecting my download and upload |
| 20:39.07 | p3nguin | I see. |
| 20:39.30 | r0m|u | so maybe not a tap issue but a comcast network issue |
| 20:39.52 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 20:40.39 | r0m|u | so far mtr has been clean. |
| 20:40.54 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 20:41.22 | p3nguin | But you said you have signal problems. |
| 20:42.03 | r0m|u | Yes. as of the last check by a tech. I dont think this issues are not related |
| 20:42.04 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 20:42.24 | r0m|u | s/are not/are/ |
| 20:45.04 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 20:45.29 | r0m|u | I really think comcast is having major issues. |
| 20:45.38 | r0m|u | beyond my small problem |
| 20:45.50 | r0m|u | I have users are work with routing problems as well from home |
| 20:45.58 | r0m|u | and they are on comcast |
| 20:46.07 | Roadblock_RVA | Comcast has been having serious issues for over a month now. At least in our area. |
| 20:46.18 | r0m|u | Roadblock_RVA: what area? |
| 20:46.33 | Roadblock_RVA | Richmond/Henrico VA |
| 20:46.38 | r0m|u | ah I see |
| 20:47.08 | Roadblock_RVA | We are seeing a lot of broken routes, and sporadic lost connectivity several times a day over the last month or so. |
| 20:47.57 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:50.36 | r0m|u | Spring Texas here and been having some major issue. |
| 20:51.31 | Roadblock_RVA | I know a little over a month ago Comcast was on the receiving end of one hell of a DOS attack. We've been seeing massive issues ever since. |
| 20:52.33 | r0m|u | Roadblock_RVA: I was not aware of that. |
| 20:54.15 | Roadblock_RVA | I don't know if or how the two are connected but the timing is pretty close for coincidence. |
| 21:03.28 | Beetlejooz | is no longer away : Gone for 43 minutes 4 seconds |
| 21:03.40 | Qwell | Beetlejooz: turn that off |
| 21:07.12 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 21:08.48 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 21:09.16 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 21:09.20 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 21:13.09 | generalhan | quick question, hopefully ... an old friend just called me for help with his asterisk system, its beeing hammered with like 20 login attempts per second, and he ask me to fix it for him. i dont have time to set up his iptables entries (which i know he has disabled) so i was trying to set it up in sip.conf using deny=0.0.0.0/0.0.0.0 and then 2 permit lines for the 1 remote user's network, and |
| 21:13.09 | generalhan | the local net, but after reloading chan_sip and doing a flat reload, you can still see the attempts flowing in ! is there something else i can try ? |
| 21:13.44 | r0m|u | generalhan: fail2ban |
| 21:14.04 | generalhan | first i tried under the [general] context in sip.conf. when that didnt work i tried under the extension they are hammering, neither worked |
| 21:14.07 | r0m|u | attempts will get there uness you do something to stop it at th border |
| 21:14.32 | [TK]D-Fender | generalhan: Of course the attempts will flow in. * can't stop packets from arriving at the server or the application. It will simply IGNORE them from non-valid IP's |
| 21:14.59 | r0m|u | [TK]D-Fender: +1 |
| 21:15.06 | generalhan | i figured the entire reason to have the deny in sip.conf was to *deny* login attempts :( |
| 21:15.32 | [TK]D-Fender | generalhan: And don;'t have time to set it up in iptables? that takes less time than you spend on your description of the problem. |
| 21:15.37 | generalhan | i dont care if his machine is being hammered ... that is not what he asked me to fix. he just doesnt want to see the failed attemps scrolling on his CLI ! lol |
| 21:15.55 | r0m|u | face palms... |
| 21:16.04 | [TK]D-Fender | Then stop them from arriving at the application level |
| 21:16.16 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 21:16.19 | p3nguin | You'll do that at the firewall. |
| 21:16.30 | IsUp | hello |
| 21:17.04 | generalhan | [TK]D-Fender: bah ... iptables will cause more issues than that ... who knows what else he has tied to this box (or the reason he isnt using iptables to begin with) i am not getting myself into that mess, maybe if i was being paid ! lol |
| 21:17.18 | p3nguin | iptables will only cause problems if you do it wrong. |
| 21:17.49 | p3nguin | You can leave the policy as ACCEPT, and just DENY those idiots that are trying to register. |
| 21:17.52 | [TK]D-Fender | generalhan: "who knows" isn't important. and it won't be doing anything other than the 2 limits you were looking to define anyway |
| 21:17.55 | p3nguin | End. Enjoy. |
| 21:18.11 | [TK]D-Fender | generalhan: If you care so little then just walk away. he clearly doesn't care about doing the job right |
| 21:18.31 | p3nguin | Give him my name. I'll fix him up correctly. |
| 21:18.47 | generalhan | [TK]D-Fender: that is very true. but i need to at least make it seem like i made an attempt! lol |
| 21:19.09 | [TK]D-Fender | Good, show his this chat. It's the neast amount of effort aside from doing nothing :) |
| 21:19.16 | generalhan | i am just going to block the IP at his router ... which will work until this person sees that he can no longer connect, and just changes his IP ! lol |
| 21:19.17 | [TK]D-Fender | least* |
| 21:19.43 | p3nguin | fail2ban will catch them when they change IP addresses. |
| 21:19.49 | p3nguin | (1513.44) <r0m|u> generalhan: fail2ban |
| 21:20.13 | generalhan | hahaha |
| 21:20.18 | p3nguin | You can run a policy of ACCEPT and still use fail2ban to block crap-heads. |
| 21:20.47 | r0m|u | * | r0m|u face palms... |
| 21:20.54 | generalhan | never heard of that ... i thought you were just being funny, saying that i failed 2 ban ! |
| 21:21.17 | p3nguin | will admin for food. |
| 21:21.22 | n3hxs | loves fail2ban :) |
| 21:21.27 | r0m|u | and again.... * | r0m|u face palms... |
| 21:24.54 | *** join/#asterisk gigawatts121 (~gigawatts@c-24-13-241-203.hsd1.il.comcast.net) |
| 21:26.28 | p3nguin | Why does freecnam fail on most numbers, but tnid has the data? They claim to use tnid for their data. |
| 21:27.53 | p3nguin | 8 out of 10 queries I've done via freecnam are blank, but when I look them up directly on tnid, there is a name. |
| 21:28.45 | [TK]D-Fender | They said they were free. Never said they were good or reliable. Try their alternate service CostsButWorksCNAM |
| 21:29.02 | p3nguin | :/ |
| 21:30.11 | gigawatts121 | Anyone know I can get Asterisk to do something with incoming Jabber messages? I have my asterisk setup as a google voice gateway, and it keeps my gmail jabber account online. I would like to log and/or route any of those incomming jabber messages to something I can see. |
| 21:30.53 | gigawatts121 | like my openfire xmpp server, or something like that |
| 21:31.25 | r0m|u | gigawatts121: you can have multiple clients running along with asterisk jabber when using GV |
| 21:31.37 | r0m|u | You have to set the priority properly |
| 21:32.17 | r0m|u | The messanger uses a prio of 24. set your asterisk to a prio of 25 |
| 21:32.24 | r0m|u | all calls will go to asterisk |
| 21:32.34 | r0m|u | and still able to see your messages on your client |
| 21:32.38 | gigawatts121 | Yeah, I know I can have multiple clients, but if all other clients are off line, and people send me messages, I would like asterisk to log them somewhere |
| 21:33.11 | gigawatts121 | If I turn verbose mode on in the asterisk console, I can see the inbound xmpp messages |
| 21:33.21 | p3nguin | You could set asterisk to a priority of 0 or -1, and the server will store the messages until a client with a positive priority signs on. |
| 21:33.29 | *** part/#asterisk Beetlejooz (~IceChat77@12.237.61.93) |
| 21:33.37 | p3nguin | No clue what that will do to voice calls, though. |
| 21:33.48 | gigawatts121 | hmm |
| 21:33.57 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 21:34.12 | r0m|u | I am sure it wont route them to asterisk |
| 21:35.06 | p3nguin | I use asterisk as a component to my openfire rather than a client. I'm not sure if that's something useful to you or not. |
| 21:35.06 | gigawatts121 | I was hoping to find a way to have it capture those inbound messages, and forward them to a user on my openfire server (where it would dump them to me when I login to that account) |
| 21:35.55 | gigawatts121 | Yeah, I tried playing around with the asterisk integration in openfire, didnt get very far |
| 21:36.58 | gigawatts121 | p3nguin: what functionality does that give you? (asterisk component in openfire) |
| 21:37.32 | p3nguin | I feel like it makes asterisk more of a peer to the server rather than acting like just another client. |
| 21:38.09 | p3nguin | I haven't gone into much depth with integration, though. |
| 21:38.39 | gigawatts121 | hmm, that sounds more along the lines of what I want to do, got any good docs/tuts about that? |
| 21:39.39 | p3nguin | Change the type from client to component in jabber.conf. Configure a component in openfire instead of a client in openfire. |
| 21:40.59 | r0m|u | p3nguin: is it possible to have a verbose set so I can see when somebody was sent to VM or called VM? |
| 21:41.13 | p3nguin | Of course. |
| 21:41.27 | p3nguin | Use Verbose() as needed. |
| 21:41.47 | r0m|u | well crap. |
| 21:41.52 | r0m|u | lol |
| 21:43.37 | j0 | I'm having random one-way audio problems. If I run a capture on the router or asterisk box I see both audio streams and everything appears to be working, but outgoing audio from my phone never makes it to the end caller |
| 21:43.54 | j0 | I have the same problem on all 4 phones here in the office |
| 21:44.30 | j0 | What would you look at first to troubleshoot it? |
| 21:44.57 | j0 | I have some data from asterisk when running sip debug, but I don't see any errors. I also don't know what I'm looking at either |
| 21:45.48 | p3nguin | Is this an intermittent problem on those four phones? |
| 21:46.05 | j0 | yes, about 1 out of every 15 calls |
| 21:46.12 | r0m|u | p3nguin: you ever seen this message? User user@gmail.com/TalkGadget1E09C3FB does not support discovery |
| 21:46.21 | p3nguin | Yes. |
| 21:46.28 | r0m|u | it pops up every so often on my asterisk |
| 21:46.35 | r0m|u | normal? |
| 21:46.57 | p3nguin | I don't know if it is normal. I don't know if I ever saw it on my own system or not. |
| 21:47.05 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 21:47.21 | r0m|u | Mhhhhhh |
| 21:47.31 | gigawatts121 | pretty sure I see it all the time |
| 21:47.45 | r0m|u | I see, |
| 21:48.08 | p3nguin | If I ever saw it on mine, it was a while ago. |
| 21:48.56 | r0m|u | I see. |
| 21:49.07 | p3nguin | and not frequently. |
| 21:49.38 | r0m|u | Is not frequent but it does popup here and there with that specific user. |
| 21:50.15 | p3nguin | That's what we call frequent. |
| 21:50.26 | p3nguin | If it happened to me, it was maybe one time. |
| 21:51.39 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:51.54 | r0m|u | p3nguin: you have other peers in your gmail list? |
| 21:52.31 | r0m|u | I am sure you dont since you only use it for asterisk. or am I wrong? |
| 21:53.22 | r0m|u | time to go home |
| 21:53.24 | r0m|u | cya! |
| 21:54.21 | p3nguin | I use my regular gmail account. |
| 21:54.38 | p3nguin | I have several people on my list on the side of the page when I go into gmail.com. |
| 21:55.10 | gigawatts121 | hmm, so with the type=component, the openfire jabber user doesnt show up as online anymore, but can still send messages to me? |
| 21:55.52 | p3nguin | It should show up online as a component. |
| 21:56.46 | gigawatts121 | as a component on the openfire admin page, yes, but that jabber user doesnt show up anymore as online in my buddy list (from yet another openfire user) |
| 21:57.43 | file | I want everyone to know that paulc is a MEANIE |
| 21:58.18 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:58.43 | gigawatts121 | would it be as simple as changing my gmail profile in the jabber.conf to be a component instead of client to not have my gmail account show up as online? |
| 22:01.15 | p3nguin | Asterisk shows up (as a component) in my roster. |
| 22:05.59 | *** join/#asterisk pdtpatrick1 (~ptaylor@12.249.4.226) |
| 22:15.07 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 22:22.07 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 22:24.59 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:25.43 | *** join/#asterisk astrutt (r0d3nt@foster.stonedcoder.org) |
| 22:33.13 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:51.27 | *** part/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 22:52.44 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 22:55.20 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-fpzvnurzolztgmug) |
| 23:10.45 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.57.16.dsl.dyn.forthnet.gr) |
| 23:16.26 | *** join/#asterisk moos3 (~rgenthner@cpe-72-224-121-41.maine.res.rr.com) |
| 23:16.44 | moos3 | is there any why to log all queries done by cdr to mysql ? |
| 23:17.30 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-kgepcrkuvraiffxk) |
| 23:24.38 | dijib | " |
| 23:25.56 | leifmadsen | moos3: that has to be done on the SQL server end, although you may or may not get some of that data with DEBUG level logging |
| 23:26.15 | moos3 | leifmadsen yeah i have binlogging in place |
| 23:26.33 | moos3 | i have in asterisk logs 95 calls and liek 30 in mysql |
| 23:26.39 | moos3 | so theres a lost of 65 calls |
| 23:26.42 | moos3 | hrm.... |
| 23:26.51 | moos3 | time for canadian beer |
| 23:26.53 | leifmadsen | but you're not seeing INSERT errors on the SQL side? |
| 23:26.58 | leifmadsen | I'm out of beer :( |
| 23:28.11 | carrar | Thats MySQL for ya |
| 23:28.38 | moos3 | i have no insert errors |
| 23:28.45 | moos3 | which is odd as hell |
| 23:29.09 | moos3 | i'm going to pull the binlogs from my pbx to my laptop and figure out why this is happening |
| 23:29.18 | moos3 | well off to beer, thanks guys |
| 23:29.25 | leifmadsen | BEER ME! |
| 23:30.13 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 23:32.13 | carrar | ONE BEER, COMING RIGHT UP!! |
| 23:39.41 | p3nguin | SAAB has announced the end of life. |
| 23:44.34 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 23:44.44 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 23:45.11 | *** part/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 23:46.22 | JerJer | can anyone remember if there is anything specific that needed to be done in Asterisk 1.4 to make it leverage multiple processors (SMP) I see Asterisk using 99% on one core, but never any others |
| 23:46.47 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 23:48.07 | p3nguin | Get an SMP kernel. |
| 23:48.15 | JerJer | got one |
| 23:48.43 | JerJer | 2.6.9-67.ELsmp #1 SMP |
| 23:48.57 | p3nguin | That should be all there is to it. |
| 23:50.16 | WIMPy | neither dahdi nor misdns echo canceller seem to be able to use more than one CPU as I found out. Are you using any of them? |
| 23:50.45 | SeRi | whats going on p3nguin |
| 23:51.45 | WIMPy | So my guess would be that this restriction applies to MeetMe as well. |
| 23:52.58 | [TK]D-Fender | or perhaps the EC module in question...."DAHDI" isn't an EC module. KB1, MG2, etc. Have you tried OSLEC or HPEC? |
| 23:53.49 | WIMPy | No, it's dahdi. It does the DSP stuff in the interrupt handling code. |
| 23:54.23 | WIMPy | Wich I think qualifies perfectly for "broken by design". |
| 23:55.28 | [TK]D-Fender | ... |
| 23:56.13 | WIMPy | That's actually the first time I feel that using an open source OS can have drawbacks. There is no one to definitely say "you must not do that". |
| 23:58.06 | coppice | There is little choice but to put the core of the EC code in the interrupt handler |
| 23:58.54 | WIMPy | There is always a choice. And Linux offers several levels. |
| 23:59.22 | coppice | maybe, but the physics of signal processing may not |
| 23:59.35 | WIMPy | There is an extra system sapce between the user and the hardware. |
| 23:59.58 | coppice | and that space incurs latency |
