| 00:02.03 | voxter | anyone experience a problem with asterisk 1.4 where when you try to forward someone a message, and use 8 to prepend your own recording, when the recipient gets it, they just hear the prepended message continuously loop and never hear the original message? |
| 00:10.41 | *** join/#asterisk sgtpepper (n=ncorrare@190.55.23.251) |
| 00:10.56 | sgtpepper | excuse me guys, my asterisk is crashing all the time |
| 00:11.09 | sgtpepper | I've an strace |
| 00:11.29 | sgtpepper | and I'm getting a very weird error |
| 00:11.30 | sgtpepper | --- SIGURG (Urgent I/O condition) @ 0 (0) --- |
| 00:11.30 | sgtpepper | rt_sigaction(SIGURG, {0x806e3c0, [URG], SA_RESTART}, {0x806e3c0, [URG], SA_RESTART}, 8) = 0 |
| 00:11.30 | sgtpepper | sigreturn() = ? (mask now []) |
| 00:11.31 | sgtpepper | read(0, <unfinished ...> |
| 00:11.31 | sgtpepper | +++ killed by SIGSEGV +++ |
| 00:11.34 | sgtpepper | that's the last |
| 00:18.39 | *** join/#asterisk VaGoNeTaS (n=dahdi@xen.datapartner.cl) |
| 00:18.42 | VaGoNeTaS | hello every1 |
| 00:19.23 | VaGoNeTaS | i got an issue |
| 00:22.47 | VaGoNeTaS | i have asterisk 1.4.24 with dahdi linux 2.2.0.2 working perfectly |
| 00:23.04 | VaGoNeTaS | but, when i upgrade asterisk from 1.4.24 to 1.4.26.1 my E1 line doesnt works |
| 00:23.24 | VaGoNeTaS | (with the same system.conf and chan_dahdi.conf files) |
| 00:23.43 | VaGoNeTaS | somebody knows something about it? |
| 00:28.47 | drmessano^ | [TK]D-Fender: You around? |
| 00:30.00 | [TK]D-Fender | drmessano^: yup |
| 00:30.18 | drmessano^ | I want to do something really ugly.. |
| 00:30.59 | [TK]D-Fender | Dirty deeds... DONE DIRT CHEAP! |
| 00:31.05 | drmessano^ | Two boxes, two diff sites, no connection between them at all.. Say I want some very ugly failover |
| 00:31.36 | drmessano^ | Basically replicate the dialplan at both sites |
| 00:31.48 | drmessano^ | No PSTN connection, SIP or otherwise |
| 00:31.58 | drmessano^ | Just ext <> ext |
| 00:32.57 | drmessano^ | I guess very short reg times would handle the connect/reconnect bit, and if my primary box came online, any existing calls on the secondary would remain until the calls end and they re-reg |
| 00:33.17 | drmessano^ | Perhaps use SRV records for weight |
| 00:33.26 | drmessano^ | What ugliness am I missing? |
| 00:33.48 | drmessano^ | Where is my idea going to go horribly wrong? |
| 00:33.57 | [TK]D-Fender | drmessano^: Phones at site A connect to server at site A normally? |
| 00:34.11 | [TK]D-Fender | drmessano^: And you want them to fail to Site B via internet if A goes down? |
| 00:34.15 | drmessano^ | All phones would be external |
| 00:35.12 | drmessano^ | Of course, any phone at A that would be internal would fail miserably with the Inet failure, if those phones should exist |
| 00:35.13 | [TK]D-Fender | drmessano^: So A & B are bother external unrelated servers? |
| 00:35.21 | drmessano^ | Yep.. |
| 00:35.44 | [TK]D-Fender | drmessano^: Well Polycom's alrady have the server.2 options |
| 00:36.02 | [TK]D-Fender | drmessano^: Or SRV is the next thought in mind |
| 00:36.04 | drmessano^ | Im thinking I could use SRV records with most phones |
| 00:36.05 | drmessano^ | Yes |
| 00:36.09 | VaGoNeTaS | so nobody knows |
| 00:37.04 | drmessano^ | So in theory, 1 goes down, and after n-time til register, they jump to box2 |
| 00:37.32 | [TK]D-Fender | drmessano^: Actually a Polycom should failover upon a failed call out as well. |
| 00:37.50 | drmessano^ | 1 serious fail is that any calls made to phone across the void would fail |
| 00:38.10 | drmessano^ | A returns, 101 jumps to A, 102 is on B, and I try to call before N-timeout |
| 00:38.39 | [TK]D-Fender | drmessano^: Life sucks.... Of course you could run a proxy at the site to maintain the routing... |
| 00:38.42 | drmessano^ | I suppose I could account for that with a peer between the boxes |
| 00:48.10 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
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| 00:51.12 | hardwire | am I insane for wanting to distribute inbound wholesale calls via app_queue? |
| 00:51.46 | *** join/#asterisk animal_ (n=animal_@201.201.187.86) |
| 00:51.57 | hardwire | granted.. it's nearly impossible for me to set up a queue with the DNIS as well |
| 00:53.23 | hardwire | sigh. |
| 00:53.26 | [TK]D-Fender | hardwire: ....huh? |
| 00:53.32 | hardwire | [TK]D-Fender: you don't want to know |
| 00:54.09 | hardwire | is freaking out because dundi doesn't handle weight like he expected. |
| 00:54.15 | hardwire | is now making a fastagi solution |
| 00:54.47 | *** join/#asterisk Whitor (n=Whitor@cpe-74-76-185-31.nycap.res.rr.com) |
| 00:58.14 | [TK]D-Fender | DUNDi is not a failover redundency solution... |
| 01:04.25 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ebcb89598b316f9f) |
| 01:11.06 | hardwire | [TK]D-Fender: Erm.. sure it is. |
| 01:11.12 | hardwire | Its just not good at it. |
| 01:11.52 | [TK]D-Fender | hardwire: Yeah, and cars aren't very good at flying either..... |
| 01:12.27 | hardwire | Yet I still want them to. |
| 01:12.52 | hardwire | [TK]D-Fender: why is the weight system not a standard weight system? If it was you could equalize distribution around like metrics. |
| 01:13.03 | hardwire | why was that one thing sorta omitted? |
| 01:14.20 | hardwire | I'll just use DNS :P |
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| 04:17.41 | manxpower | . |
| 04:18.48 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 04:20.11 | [TK]D-Fender | manxpower: I see your point |
| 04:20.22 | [TK]D-Fender | manxpower: and raise you 3... |
| 04:38.25 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:38.30 | joelsolanki | HI all. |
| 04:39.03 | joelsolanki | i have installed asterisk 1.4.x and it says it has t.38 fax support. we tried to send fax but it is not working. |
| 04:39.15 | [TK]D-Fender | joelsolanki: T.38 PASSTHROUGH |
| 04:39.19 | joelsolanki | then we tried fax over g711 but also not working. |
| 04:39.29 | joelsolanki | yes t.38 pass thru |
| 04:40.11 | joelsolanki | asterisk-1.4.X --> voip provider --> client fax machine will this work ? |
| 04:40.27 | [TK]D-Fender | joelsolanki: that says nothing about the EQUIPMENT involved |
| 04:40.39 | [TK]D-Fender | joelsolanki: So far "fax machine" sure doesn't say T.38 to me. |
| 04:41.05 | joelsolanki | it is traditional fax equipments. |
| 04:41.09 | joelsolanki | so it must be t.38 |
| 04:41.24 | [TK]D-Fender | [00:40]<joelsolanki>asterisk-1.4.X --> voip provider --> client fax machine will this work ? <- and this looks like ASTERISK sending a fax. That would not be "passthrough" |
| 04:41.45 | joelsolanki | oh no |
| 04:41.49 | [TK]D-Fender | joelsolanki: T.38 is not a protocol on an ANALOG FAX MACHINE |
| 04:41.52 | joelsolanki | let me again desribe |
| 04:42.04 | [TK]D-Fender | joelsolanki: Yes, please do, you are getting worse by the moment |
| 04:42.12 | joelsolanki | :) |
| 04:42.23 | [TK]D-Fender | Traditional fax machines know nothing about the internet |
| 04:42.41 | dupe | please let fax machines die! |
| 04:42.43 | dupe | sobs |
| 04:42.46 | joelsolanki | analog phone --> zyxcel IP DSLAM --> asterisk 1.4.X --> voip provider --> customer fax machine |
| 04:42.57 | [TK]D-Fender | waits for it to pool, then holds dupe's head under |
| 04:43.07 | dupe | that sounds like my last vacation... |
| 04:43.39 | [TK]D-Fender | joelsolanki: DSLAM? pardon? What the hell does DSL have to do with this? And since when does an analog PHONE send a fax? |
| 04:43.46 | joelsolanki | in that analog phone fax machine is connected |
| 04:44.19 | dupe | asterisk doesnt RECEIVE faxes if thats what you're asking |
| 04:44.22 | dupe | or SEND them |
| 04:44.26 | [TK]D-Fender | [00:42]<joelsolanki>analog phone --> zyxcel IP DSLAM --> asterisk 1.4.X --> voip provider --> customer fax machine <-- there is only 1 fax machine mentioned here. |
| 04:44.35 | joelsolanki | no customer wants to send fax out. |
| 04:44.35 | [TK]D-Fender | dupe: it can with added apps |
| 04:44.45 | joelsolanki | 1 sec |
| 04:44.52 | dupe | heh well yeah but not how he wants to do it i think.. |
| 04:44.59 | dupe | im trying to keep things simple |
| 04:45.14 | dupe | of course i could just be wrong. thats happened before. |
| 04:45.29 | dupe | life is so much better after i get a good 13 hour nap. |
| 04:45.47 | joelsolanki | fax machine --> zyxcel IP DSLAM --> asterisk 1.4.X --> voip provider --> remote fax machine |
| 04:46.02 | [TK]D-Fender | joelsolanki: What is this "DSLAM" doing? |
| 04:46.08 | joelsolanki | we are providing digital phone services therefore zyxcel IP DSLAM is in picture. |
| 04:46.15 | [TK]D-Fender | joelsolanki: and your PROVIDER must support T.38 as well |
| 04:46.31 | joelsolanki | my provider support T.38 passthru. they use quintum. |
| 04:46.35 | [TK]D-Fender | joelsolanki: What does it talk to *? |
| 04:46.43 | joelsolanki | yes |
| 04:46.49 | joelsolanki | asterisk is connected to quintum. |
| 04:46.56 | joelsolanki | using IP. |
| 04:47.13 | [TK]D-Fender | joelsolanki: ....... |
| 04:47.39 | [TK]D-Fender | joelsolanki: Redescribe that shit and stop doing it in PIECES and confirm what PROTOCOLS are supported at each step |
| 04:47.57 | joelsolanki | ok |
| 04:50.26 | joelsolanki | we are providing digital phone services so our customers has analog phone lines means copper lines. copper lines are connected to our DSLAM which is supporting SIP protocol. In DSLAM we setup a each account and it gets connected to our asterisk server using IP. |
| 04:50.32 | joelsolanki | Fax machine --> DSLAM --> Asterisk 1.4.X --> Quintum(Provider) --> Remote fax machine |
| 04:50.47 | [TK]D-Fender | ARGH |
| 04:51.23 | dupe | joelsolanki: does your dslam equipment etc support t38 passthrough? |
| 04:51.28 | [TK]D-Fender | joelsolanki: DSLAM <- what fucking PROTOCOLS is this thing talking to *? SHOW ME that it supports T.38 |
| 04:51.39 | joelsolanki | yes dslam supports t.38 pass thru |
| 04:51.56 | dupe | fax = t38 you need to ensure every part supports it.. |
| 04:51.59 | dupe | oh |
| 04:52.01 | [TK]D-Fender | joelsolanki: FINE. About F-ing time. |
| 04:52.19 | [TK]D-Fender | joelsolanki: So yes, * can allow that to pass through once configured properly |
| 04:52.21 | joelsolanki | DSLAM is using SIP protocol to connect to asterisk |
| 04:52.40 | [TK]D-Fender | joelsolanki: now that is not saying it is going to WORK. T.38 interoperability is not guaranteed |
| 04:52.55 | joelsolanki | oh. |
| 04:52.58 | dupe | faxing is so 90s anyway :P |
| 04:53.07 | dupe | faxing is iffy on best of days... |
| 04:53.24 | joelsolanki | yes i understand. but u know customers asks so we have to provide it. |
| 04:53.40 | joelsolanki | i am also going to test fax over g711 |
| 04:54.24 | joelsolanki | i know it is also not best solution but i will checkout how much sucess ratio it gives |
| 05:01.16 | [TK]D-Fender | dupe: My favourite part of that is all we got was "it didn't work", No error messages. No SIP debug show negotiation. No configs to prove he actually tried to configure it at all VS imagining that it all must happen "like magic". NOTHING |
| 05:01.47 | dupe | asterisk is magic |
| 05:01.51 | [TK]D-Fender | dupe: And you have to beat him over the head to get a simple linear description of whats involved. |
| 05:01.53 | Qwell | SIP on a DSLAM? |
| 05:02.24 | [TK]D-Fender | Qwell: NFC <- |
| 05:02.26 | dupe | i'm surprised you guys dont have triggers setup that automatically say $NICK no, answer my question you dumbass! |
| 05:02.30 | dupe | instead of repeating it :D |
| 05:02.46 | [TK]D-Fender | Qwell: I still want to know WTF DSl has to do with this. |
| 05:03.24 | [TK]D-Fender | Qwell: I mean he plug a fax into it and it talks SIP to *. Sounds like an F-ing ATA to me... |
| 05:03.25 | dupe | i got the impression it was a local inhouse dslam, with a private backhaul into "the internet" |
| 05:03.33 | Qwell | yeah... |
| 05:04.06 | [TK]D-Fender | A description right out of CrazyWorld |
| 05:04.41 | dupe | but you're very patient so you get coffee+cookies :) |
| 05:04.56 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
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| 05:06.24 | [TK]D-Fender | dupe: I'll save'em for tomorrow... its bed time... |
| 05:07.06 | [TK]D-Fender | dupe: Actually tomorrow I'm off for a AYCE sushi lunch.... don't think I'll be eating anything ftill then..... and probably nothing more that day following |
| 05:07.48 | dupe | sushi? ewwww that killed my appetite. thats the one few things i just cant handle |
| 05:09.33 | [TK]D-Fender | <- Omnivore |
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| 05:41.13 | keebler | Has anyone used the Patton M-ATA? |
| 05:52.26 | kb3ien | not me |
| 05:54.17 | keebler | just currious of the call quality in comparison to the Linksys PAP2 |
| 05:54.27 | keebler | And power consumption |
| 05:54.42 | keebler | I'd really like to run it off 3vdc |
| 05:58.25 | *** join/#asterisk fiddur (n=fiddur@192.121.104.122) |
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| 06:06.24 | kb3ien | 3volts dc must mandate some very short or thick wiring, no? |
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| 06:08.31 | fiddur | Hi folks. I have a problem with realtime sip peers. It seems like the expiration timer kicks in even if the device has re-registered. E.G: A device registeres 06.25 with expiration of 3600 seconds. It registeres again at 07.24, but 07.25 it gets unregistered by expire. I know there is a kind of "realtime caching", but that doesn't seem like the correct solution. |
| 06:17.03 | *** join/#asterisk xrmx__ (n=rm@host204-254-dynamic.15-87-r.retail.telecomitalia.it) |
| 06:21.59 | *** join/#asterisk justanothercoder (n=devil@121.243.137.253) |
| 06:26.36 | justanothercoder | hello, i'm using asterisk and have everything setup, am using perl agi, is there any reason why say_digits works, but say_alpha doesn't work? |
| 06:27.07 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 06:31.23 | Dovid | i use php+agi and I use the exec command and put it in there |
| 06:32.12 | justanothercoder | i guess its the same with perl too, but for some reason, i don't hear anything when i use say_alpha |
| 06:32.31 | justanothercoder | no error, nothing in the debug logs, its as if the line doesn't exist |
| 06:32.41 | justanothercoder | i tried both say_alpha and say_phonetic |
| 06:33.08 | *** part/#asterisk kron4eg (n=kron@unaffiliated/fixer/x-387521) |
| 06:33.25 | Dovid | i had issues in php agi with some of the says so i just used exec |
| 06:34.20 | justanothercoder | how do you exec? |
| 06:34.43 | justanothercoder | could you please paste the command? |
| 06:38.03 | fiddur | Has anyone gotten realtime sip peers to work? They expire even though they were re-registered.. see here: http://pastebin.com/m627b91c That's debug/verbose and the manager events PeerState that shows on unregister. I don't understand though - the debug-log says "devicestate.c: Changing state for SIP/124 - state 1 (Not in use)" even on unregister; that would mean it is available? ...but I can't call it, and it's not in sip autoreg... |
| 06:39.29 | *** join/#asterisk maour (n=quassel@unaffiliated/maour) |
| 06:41.18 | maour | I edit extensions.conf and then i want to reload it but in CLI> there is no command with the name of "extension reload" , why ?! ( asterisk 1.6.1.4) |
| 06:41.30 | justanothercoder | reload |
| 06:41.35 | justanothercoder | just type reload |
| 06:41.48 | fiddur | or dialplan reload |
| 06:41.50 | justanothercoder | @maour |
| 06:42.05 | maour | and where is extension reload ? :-/ |
| 06:42.14 | kaldemar | maour: it has been removed in 1.6 |
| 06:42.19 | maour | o |
| 06:42.21 | maour | thanks |
| 06:42.59 | kaldemar | so dialplan reload is the equivalent now, like fiddur said. no need to reload everything with "reload". |
| 06:44.27 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 06:45.11 | maour | when is just type reload i get this errors and warnings http://pastebin.ca/1527640 |
| 06:45.25 | maour | but dialplan reload wokrs fine |
| 06:46.51 | maour | works* |
| 06:47.19 | justanothercoder | Dovid : can you please paste the exec command here? |
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| 06:51.53 | kaldemar | maour: reload tries to reload configuration for every module you have in asterisk. |
| 06:52.17 | maour | ok , then no need to it |
| 06:54.11 | maour | what about " CLI> show dialplan 6599@features " ? seems i should type "CLI> dialplan show" but how can i specify a specific extension ? |
| 06:55.47 | fiddur | maour: `help dialplan show` |
| 06:56.01 | maour | oh! i forgot about help |
| 06:57.54 | maour | is there any reference about commands that changed since 1.4 to 1.6 ? |
| 07:03.33 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 07:04.53 | kaldemar | maour: http://svn.digium.com/svn/asterisk/tags/1.6.1.4/UPGRADE-1.6.txt |
| 07:05.29 | kaldemar | maour: http://svn.digium.com/svn/asterisk/tags/1.6.0.13/CHANGES |
| 07:05.34 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 07:05.57 | maour | kaldemar: thanks . |
| 07:07.20 | kaldemar | CHANGES in 1.6.1 packages have similar changes between 1.6.0 and 1.6.1. |
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| 07:53.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:55.20 | maour | when i press tab twice in CLI> it shows so many command that i can type , for example i have restart command but when i type restart it says No such command 'restart'. !! (it also complete it for me when i just type res and then press tab) |
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| 07:58.02 | Tim_Toady | type help restart as it asks you to do, or type restart and press tab-tab again |
| 08:01.06 | maour | help restart |
| 08:01.06 | maour | No such command 'restart'. |
| 08:01.12 | maour | Tim_Toady: ^ |
| 08:01.47 | Tim_Toady | what asterisk version? |
| 08:02.01 | maour | 1.6.1.4 |
| 08:02.25 | maour | but when i type restart and i press tab twice i get gracefully now when |
| 08:02.29 | maour | seems ok |
| 08:06.06 | kaldemar | maour: core help restart will give you the helps |
| 08:06.49 | maour | No such command 'core help restart' (type 'help core help' for other possible commands) |
| 08:06.52 | maour | kaldemar: ^ |
| 08:08.22 | *** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) |
| 08:08.29 | LtScarr | hey everybody |
| 08:08.39 | Tim_Toady | sems like somehting is wrong with help command in 1.6.1 :P |
| 08:09.38 | LtScarr | i have a question regarding queues and not letting it ring busy members |
| 08:09.57 | fiddur | maour: try help core restart |
| 08:09.57 | fiddur | help first on the line... |
| 08:11.01 | maour | fiddur: now i got something :p |
| 08:13.15 | kaldemar | maour: oops. help core restart |
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| 08:30.49 | Stese | Morning all.... |
| 08:31.53 | Stese | I've an issue with Audio between a phone and an asterisk box. Sip is fine, with Nat... but audio isn't getting though,,, |
| 08:32.41 | Stese | I'm guessing it's becuase the RTP packets are getting lost, but would someone be able to advise where I need to look to fix this? |
| 08:33.16 | Stese | ie if it's more likely an asterisk setting, or a router issue |
| 08:33.37 | LtScarr | Stese: check your canreinvite settings |
| 08:34.16 | LtScarr | setting this to no fixes most problems |
| 08:34.31 | kaldemar | ~sipnat |
| 08:34.31 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 08:34.45 | Stese | thanks kaldemar :) |
| 08:34.53 | Stese | i'll flick though that.... :) |
| 08:35.35 | Stese | LtScarr... my sip.conf has nothing in there at the moment (I started with a clean file to prevent me getting confused) I'm guessing I shuold be allowing reinvites |
| 08:37.02 | LtScarr | Stese: yes you need to add that line yourself |
| 08:37.30 | LtScarr | you can put it in the general section |
| 08:37.34 | LtScarr | or per peer |
| 08:37.49 | Stese | LtScarr > am I right in thinking that allowing reinvites will break MixMonitor() recording? |
| 08:38.00 | Stese | or have I got very confused? |
| 08:38.11 | LtScarr | Stese: i have no idea actually :) |
| 08:38.36 | LtScarr | but you should try it if you are in a test environment |
| 08:39.23 | Stese | Well, it was due to go live yesterday (it didn't work), but they have phones, so i'm still testing... i'm not lucky enough to have a test enviroment |
| 08:40.29 | LtScarr | let me know if it works out |
| 08:40.46 | LtScarr | it solved my many nat problems |
| 08:40.50 | Stese | I will try to remember :) |
| 08:41.10 | Stese | i'm going to head to site this afternoon, so i'll be live from there |
| 08:42.44 | Stese | annoyingly enough, the phone has stopped registering, so i've got that to fix too |
| 08:43.00 | LtScarr | i know that feeling :) |
| 08:43.44 | Stese | it was workign yesterday |
| 08:46.29 | Stese | if i'm honest, I don't like this set up at all |
| 08:46.41 | Stese | much prefer asterisk and the phones on the same site |
| 08:46.56 | LtScarr | ouch indeed :D |
| 08:47.31 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 08:47.41 | Stese | but as usual, it's been sold to the customer, without any time or thought to it being workable |
| 08:52.02 | Stese | Hmm... the phone keeps going unreachable, dispite a good ping time, dammit |
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| 08:58.07 | Stese | hmm, making the phone do nat keepalives seems to fix that *phew* |
| 08:58.32 | Stese | canreinvite=yes hasn't made a difference to the audio |
| 09:00.26 | LtScarr | Stese: where did you put that in? |
| 09:01.20 | kaldemar | Stese: qualify parameter is something that can be used with nat too. |
| 09:01.24 | Stese | [General] In sip.conf |
| 09:01.33 | LtScarr | Stese: it's canreinvite=no |
| 09:01.36 | LtScarr | not yes :) |
| 09:01.42 | Stese | oh... *blush* |
| 09:02.07 | Stese | kaldemar > qualify=yes at the moment, |
| 09:03.02 | Stese | canreinvite=no makes no difference |
| 09:03.13 | Stese | i have a feeling it will be the local router |
| 09:03.19 | mort_gib | Stese: look at your firewall settings |
| 09:03.20 | Stese | gonna get a trace from it |
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| 09:13.52 | LtScarr | Stese: have you specified the nat setting? |
| 09:14.30 | Stese | for the phone in the sip.conf? yes |
| 09:14.33 | Stese | nat is on |
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| 09:52.28 | zeeesh | anybuddy there ? |
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| 10:02.15 | Stese | I am |
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| 10:14.59 | zeeesh | sip to gtalk .. connection established, but no voice at both side .. nat=yes in sip.conf General secotion as well in sip peer account, canreinvite=no in both general and sip peer account, ? what could be the probability there is no voice at both end? |
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| 10:25.34 | zeeesh | <Stese>: u there ? |
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| 10:31.01 | Stese | the probability is 1, since you don't have audio. |
| 10:32.02 | Stese | I'm having a similar issue, with SIP/RTP atm, you've looked at everything I have so far... the common feeling it is a firewall/routing issue |
| 10:34.08 | *** join/#asterisk Jabka (n=jabka@hspot.sce.ac.il) |
| 10:34.43 | Jabka | What is the prefered way to call alpha numeric client on diffrent servers ? is it dundi or some other option ? |
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| 10:35.54 | Jabka | im searching for some solution so user that registered to one server would be able to call other servers (as with sip proxy) , but not with sip trunks (since sip trunk afaik will not forward the caller info) |
| 10:40.54 | tzafrir_laptop | Jabka, hi :-) |
| 10:41.15 | Jabka | tzafrir_laptop , Hi :-) |
| 10:41.35 | Jabka | tzafrir_laptop , i hope the creator of sebi channel implementated my notes |
| 10:41.55 | tzafrir_laptop | well, asterisk is not a sip proxy. Maybe try using a sip proxy (e.g. one of the SERs) |
| 10:42.34 | tzafrir_laptop | resolve a dialed number to some voip address: that would be ENUM, I guess |
| 10:42.47 | Jabka | tzafrir_laptop , i intend to use what i know (OpenSips) to be able to work as a sip proxy but i wish to use a known and defined way so there will be as less as posible changes |
| 10:43.23 | kaldemar | Jabka: DUNDi is for locating numbers, not for calling. |
| 10:43.26 | Jabka | ENUM afaik using a heirarchy model and it isn't the roubustest way |
| 10:43.55 | Jabka | kaldemar , so DUNDi is out of the picture ? |
| 10:44.48 | Jabka | the issue is that i don't want to change lots of code to make it work (tzafrir you remember how many changes i had to do for OpenSips to make it work on Farms) |
| 10:44.52 | kaldemar | i'm not completely sure i understand what you aim for, just stated that DUNDi is a number discovery protocol and not used for call setup |
| 10:45.52 | Jabka | kaldemar , the idea is it provide a service (network ? ) that will alow a server (asterisk) to make "a call" to diffrent server but also provide the abiulity to call directly back |
| 10:46.37 | Jabka | for instance foo@asterisk.example.com call bar@someotherAsterisk.com and back |
| 10:47.15 | Jabka | I used Alert system in OpenSer for meetings but Asterisk is much isear for this (meetMe) |
| 10:51.01 | zeeesh | is there any feature of realtime for jabber.conf? |
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| 10:53.59 | kaldemar | Jabka: what might cause loop troubles with asterisk |
| 10:54.52 | Jabka | kaldemar: could you put some light on it ? do you meen loose route ? |
| 10:58.17 | kaldemar | if bar@someotherAsterisk.com doesn't start a new call leg to foo@asterisk.example.com, asterisk might think it's a loop |
| 10:59.48 | *** join/#asterisk sublimerix (n=sublime@81.199.146.125) |
| 11:00.41 | Jabka | the intetition that you will call only to server that support the service |
| 11:01.27 | sublimerix | ok |
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| 11:02.27 | Jabka | kaldemar , btw when you say call leg you meen only the SIP part ? (or it can involve connection to PSTN ? ) |
| 11:02.52 | kaldemar | i meant the SIP part |
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| 11:03.45 | *** part/#asterisk sublimerix (n=sublime@81.199.146.125) |
| 11:03.46 | kaldemar | DUNDi can be used to query multiple servers for a number. if a server has that number in the used DUNDi mapping, it will answer with a URI used for dialing the number. |
| 11:03.54 | Jabka | kaldemar , whell afaik it is easy to test that (that you make sip calls) the problem is with PSTN connections |
| 11:04.58 | Jabka | so i guessed right DUNDi can be used for that , gr8 (need to study much more about it) |
| 11:05.07 | kaldemar | so, it you prefix the service (a PSTN connection in this case, i assume) with a certain number, you could use DUNDi to discover the service. |
| 11:06.08 | Jabka | i intened not to use in a prefix but using the realm ( i have an implementation for that) |
| 11:07.10 | Jabka | moment can i dial to diffrent realms from sip phones ? (last one i had was very long ago) |
| 11:07.13 | Jabka | ? |
| 11:07.51 | box2 | welcome to pimsleur's eastern arabic 1 |
| 11:08.13 | box2 | in the next few minutes, you will not only learn to understand this conversation, but to take part in it as well |
| 11:08.41 | Jabka | ha ? |
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| 11:29.19 | *** join/#asterisk kev_taff (n=kevin@poirot1.pck.nerim.net) |
| 11:29.23 | kev_taff | hello |
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| 11:30.02 | kev_taff | anyone can tell me why tos=0x10 is deprecated in sip.conf please? |
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| 11:36.48 | obruT | does anybody knows difference between cdr_adaptive_odbc and cdr_odbc ? I just run make menuselect, cdr_adaptive_odbc is available, but cdr_odbc is not ?! |
| 11:37.26 | kaldemar | kev_taff: it was divided into tos_sip, tos_audio, tos_video and tos_text |
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| 11:45.18 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 11:46.09 | arekm | hi, will asterisk built with dahdi work with zaptel kernel drivers? Or do I need dahdi kernel drivers, too? (I'm using some weird drivers that are modified zaptel stuff, so switching to dahdi drivers will be a problem) |
| 11:47.07 | tzafrir_laptop | arekm, generally: no. But what version of asterisk it is? |
| 11:49.19 | arekm | tzafrir_laptop: 1.4.21 currently, could switch to 1.6 |
| 11:49.20 | tzafrir_laptop | arekm, and btw: http://svn.asterisk.org/svn/zaptel/branches/1.4/build_tools/cvt_zapdahdi might be of some use to you |
| 11:49.47 | tzafrir_laptop | 1.4.21 does not support dahdi. 1.6.0 and on do not support zaptel at all |
| 11:50.55 | arekm | and I want to switch to 1.6... looking at that script. If it's just matter of changing macros and function names in driver then it shouldn't be a problem |
| 11:51.49 | KazaLite | can someone tell me when we make a call, which function in asterisk decides which codec to be used? |
| 11:52.20 | tzafrir_laptop | there's generally one tricky point - chans in the dahdi_span struct was changed: |
| 11:52.47 | tzafrir_laptop | it's an array of pointers to channels rather than an array of channels |
| 11:53.12 | tzafrir_laptop | That is the major reason for oopses when converting drivers to dahdi |
| 11:55.42 | kev_taff | kaldemar: yes, so, i've the same warning if i use tos_* |
| 11:55.57 | kev_taff | it's the value "0x10" who is deprecated |
| 11:58.59 | kaldemar | see doc/tex/qos.tex in the source package |
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| 12:29.47 | unasi7 | is there a festival (TTS) guru? |
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| 12:30.50 | ariel_ | Morning |
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| 12:45.07 | dandre | hello, |
| 12:45.49 | dandre | I still have problems displaying callerid on my analog phone connected to fxs tdm400 port. |
| 12:45.57 | dandre | I have found this: |
| 12:45.58 | dandre | http://www.voipuser.org/forum_topic_3727.html |
| 12:46.32 | dandre | is it true that "the TDM400 FXS ports apparently only support US caller ID, unlike the FXO port" ? |
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| 12:49.24 | SuPrSluG | what would cause one way audio DURING a call? It doesn't last long, but does occur. My guess would be network jitter or bandwidth. Anyone else experience anything like this? |
| 12:49.56 | manxpower | packetloss, or *major* jitter. |
| 12:50.33 | SuPrSluG | manxpower: any good tools to test network jitter? |
| 12:50.51 | manxpower | I really wish there was an option for Windows Update to set "don't ever reboot my computer in the middle of the night when I have important files open" |
| 12:51.08 | SuPrSluG | lol |
| 12:51.12 | [TK]D-Fender | manxpower: There is. |
| 12:51.24 | manxpower | SuPrSluG: ICMP (which ping uses) should always have the same or worse jitter than audio. |
| 12:51.46 | manxpower | [TK]D-Fender: I switched to "just notify me of updates", but that does not always seem to work. |
| 12:51.51 | [TK]D-Fender | manxpower: http://www.aeonity.com/frost/remove-automatic-restart-after-windows-updates |
| 12:53.09 | manxpower | [TK]D-Fender: if that works, you may get a marriage proposal. |
| 12:55.56 | dandre | is it true that "the TDM400 FXS ports only support US caller ID, unlike the FXO port" ? |
| 12:56.22 | manxpower | dandre: that is a question for Digium support. |
| 12:56.52 | dandre | ok how should I reach them? |
| 12:57.00 | [TK]D-Fender | dandAnd don't repeat your questions every 10 minutes because nobody had anything to contribute |
| 12:57.36 | [TK]D-Fender | dandre: www.digium.com <- I'm sure they list a half dozen ways... also have you read the EXTENSIONS.CONF SAMPLE? |
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| 12:59.47 | tzafrir_laptop | dandre, that claim is a bit strange. with FXS you can provide caller ID. With FXO you can read caller ID |
| 13:00.19 | tzafrir_laptop | (assuming standard signalling as implied from "US caller ID") |
| 13:00.52 | dandre | [TK]D-Fender: yes I have read it but see nothing related to my problem |
| 13:01.16 | manxpower | tzafrir_laptop: It's an asterisk web forum -- it's users are the blind leading the blind. |
| 13:01.25 | [TK]D-Fender | dandCONTACT NUMBERS. Please actually loko at what I'm responding to |
| 13:01.33 | [TK]D-Fender | dandre: CSo go CALL THEM. |
| 13:01.45 | dandre | ok |
| 13:02.52 | tzafrir_laptop | dandre, most phones will do a pretty good job at detecting caller IDs in most common protocols |
| 13:03.17 | tzafrir_laptop | And anyway, if you send them a proper US (bell) caller ID, it should be detected |
| 13:03.58 | dandre | apparently it doesn't on my phone |
| 13:05.14 | dandre | I think that the callerid protocol in france isn't the same as in us |
| 13:05.44 | dandre | on my phone there is a tag "design for France" |
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| 13:23.55 | *** join/#asterisk moa_ (n=moa_@65-19-228-168.vnet-inc.com) |
| 13:24.12 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 13:29.21 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:31.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:35.16 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
| 13:35.33 | RypPn | dandre maybe this will help, http://pastebin.com/m768f80b1 can't hurt to give it a try |
| 13:36.56 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:37.03 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:37.32 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 13:38.41 | Katty | :> |
| 13:39.46 | manxpower | RypPn: that does not help at all. |
| 13:40.05 | manxpower | His problem is with an FXS port, the pastebin you provided is for FXO ports. |
| 13:41.30 | dandre | RypPn: I have done that but doesn't work |
| 13:41.52 | dandre | I still use zap channels instead of dadhi |
| 13:41.55 | *** join/#asterisk propellerhead (n=yogurt2u@host112.190-30-35.telecom.net.ar) |
| 13:42.58 | tzafrir_laptop | 'callerid=asrecieved' is for the FXO. dandre configures an FXS adapter |
| 13:43.19 | tzafrir_laptop | oh, and someone already mentioned it... |
| 13:43.33 | manxpower | tzafrir_laptop: I guess some people think FXO and FXS are the same thing. |
| 13:44.10 | RypPn | nice backhander |
| 13:44.18 | manxpower | The sames one that think NO2 is the same thing as N2O, I bet. |
| 13:44.35 | tzafrir_laptop | now that's funny |
| 13:45.20 | *** join/#asterisk danfowler (n=DFowler@ip-66-9-231-226.autorev.intellispace.net) |
| 13:46.37 | dandre | I have tried everything :-( |
| 13:46.59 | manxpower | dandre: did you try calling Digium? |
| 13:47.27 | dandre | no because I don't speak english |
| 13:48.02 | manxpower | Then you have not done the one thing that will get you the correct answer. E-mail Digium if you cannot call them. |
| 13:48.05 | eppigy | hello |
| 13:48.06 | eppigy | i am dave |
| 13:48.19 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:52.02 | Katty | eppigy: :> |
| 13:52.09 | Katty | hugs on eppigy |
| 13:52.13 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 13:53.29 | AndyML | where'd all teh griefer's come from? |
| 13:53.30 | dandre | ok |
| 13:55.08 | *** join/#asterisk coppice (n=chatzill@26.168.17.210.dyn.pacific.net.hk) |
| 13:55.31 | eppigy | surrenders completely to Katty |
| 13:55.42 | blkry | can someone tell me were to look to troubleshoot this error?: [2009-08-13 09:50:40] WARNING[24025] chan_sip.c: Remote host can't match request BYE to call '7d556c6a0d93e33b3c75dee74ddc1454@192.168.1.100'. Giving up. it happens every 7 seconds in full.log. |
| 13:59.28 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:04.41 | gego | Hi. Can anyone help me with this problem: Our doorcom calls * through an analogue line via ATA. |
| 14:05.26 | gego | I can make any phone ring - but only the grandstreams can send DTMF inband. |
| 14:06.26 | gego | So I can only configure these phones (which I don't like at all) to respond to the intercom and open the door. |
| 14:07.16 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:07.16 | manxpower | gego: I have no idea what you just said. |
| 14:07.23 | manxpower | inband DTMF is not a problem. |
| 14:07.38 | manxpower | (not a problem for Asterisk) |
| 14:08.25 | gego | OK - I just don't know how to NOT bridge certain calls. |
| 14:08.42 | gego | The fax is on the same ATA and needs bridging |
| 14:09.01 | *** join/#asterisk puzzled (n=foobar@188.90.134.167) |
| 14:09.34 | gego | that's why I was talking about the phones. |
| 14:09.56 | n3hxs | gego look at the last post at http://www.voipcoop.org/viewtopic.php?t=70 |
| 14:11.50 | *** join/#asterisk rnovotny22 (n=Bob@75-161-133-212.mpls.qwest.net) |
| 14:12.25 | manxpower | gego: look at the Wait and WaitForRing applications. |
| 14:17.02 | gego | OK - thanks. But I can't see what that's good for? My Problem is that I'm the callee. Otherwise SIPDtmfMode would do ?!? |
| 14:17.40 | manxpower | You are not explaining the problem well. It seems like you have a door phone and a fax on the same ATA. Your door phone calls Asterisk, but the fax is answering the line. |
| 14:17.45 | manxpower | you set the dtmfmode in sip.conf |
| 14:18.04 | manxpower | or maybe someone tries to send you a fax and Asterisk answers. |
| 14:19.26 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 14:19.56 | manxpower | Regardless of all of this if you need to change the DTMF mode, then set it in sip.conf [section] for that device and then set the mode on the actual device. |
| 14:21.42 | manxpower | I would not be surprised if the Grandstreame device does not support playing DTMF on it's FXS port. Many such devices don't support that. |
| 14:21.44 | manxpower | !phones |
| 14:21.46 | manxpower | ~phones |
| 14:21.47 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 14:24.12 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 14:29.45 | blkry | Has anyone used one of the new Cisco SPA5xx? |
| 14:33.04 | *** join/#asterisk jtodd (i=du9qkt2d@ns.fox-den.com) |
| 14:33.04 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:36.35 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 14:38.56 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 14:39.00 | ayeso | Need some help, I am running version 1.6.1.0 all SIP, "sometimes" RTP doesn't work, signaling is always good... Im not sure what is going on here. |
| 14:40.06 | Katty | ayeso: i've had that before, sometimes. |
| 14:40.16 | Katty | ayeso: usually i found that my rtp port range in the firewall wasn't large enough. |
| 14:40.27 | Katty | ayeso: so some packets would go through properly, and some would trigger a policy violation |
| 14:41.31 | ayeso | Katty: no firewall involved here, inbound/outbound is SIP to a cisco 3825 media gateway, the gateway has PRIs |
| 14:41.48 | Katty | good luck (= |
| 14:42.09 | *** part/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 14:43.01 | *** join/#asterisk moy (n=moy@74.12.123.137) |
| 14:45.17 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 14:45.35 | gego | manxpower: sorry for my confused description! |
| 14:47.02 | *** join/#asterisk tengulre (n=tengulre@118.115.134.138) |
| 14:47.04 | gego | the ATA is on one port on an 8-port BRI-ISDN card. It's configured in misdn.conf |
| 14:47.09 | Faustov | tzafrir_laptop, [TK]D-Fender ping |
| 14:47.30 | Faustov | got my issue solved in you're curious |
| 14:47.38 | tzafrir_laptop | is |
| 14:48.31 | Faustov | so the guy from sangoma logged in |
| 14:48.37 | Faustov | removed everything |
| 14:48.38 | gego | and the ATA makes it 2 a/b ports for door and fax device. |
| 14:48.48 | Faustov | recompiled from source instead of gentoo's portage |
| 14:48.50 | Faustov | and it worked |
| 14:48.53 | Faustov | cries |
| 14:49.06 | box2 | also cries |
| 14:49.26 | beek | just sniffles. |
| 14:49.37 | Faustov | apparently downgrade to *-1.4 and then upgrade back to *-1.6 left 2 versions of dahdi-linux running and fighting for precedence |
| 14:49.40 | Katty | aww |
| 14:49.42 | Katty | no sniffles :< |
| 14:49.47 | Katty | distributes hugs, tissues, and chocolate |
| 14:49.54 | Faustov | accepts |
| 14:50.03 | beek | greedily snaps up two of the chocolate |
| 14:50.04 | Faustov | but yay, it works now, hurray! |
| 14:51.43 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 14:52.47 | Katty | :> |
| 14:53.04 | [TK]D-Fender | GENPOO |
| 14:54.32 | beek | 'morning gang |
| 14:54.45 | *** join/#asterisk scalex000 (n=chatzill@36puntacana02.codetel.net.do) |
| 14:54.58 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:54.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:56.54 | *** join/#asterisk s14ck (n=s14ck@190-76-119-198.dyn.movilnet.com.ve) |
| 14:59.12 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:04.04 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:05.59 | *** join/#asterisk PuroOsso (n=PuroOsso@189.74.141.10) |
| 15:08.37 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:14.23 | box2 | http://www.securityfocus.com/archive/1/505723 |
| 15:15.41 | [TK]D-Fender | \o/ |
| 15:16.50 | Naikrovek | is wondering if it's just him... |
| 15:17.08 | Naikrovek | anyone ever have network issues installing AsteriskNOW on vmware? |
| 15:23.29 | Faustov | [TK]D-Fender: I knew you'd say something like that... |
| 15:23.30 | Faustov | :> |
| 15:26.52 | Katty | http://www.thinkgeek.com/geektoys/6279/ <- is this an acceptable christmas present for an ex military computer techy geek? |
| 15:28.52 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 15:29.34 | box2 | shrugs |
| 15:29.37 | grandpapadot | Naikrovek: I had issues with 1.2 on debual 3.x with vmware server 1.0, seems like everything was jittery. No issues with debian 4 or 5 and asterisk 1.4 on vmware workstation 6. |
| 15:29.53 | grandpapadot | s/debual/debian |
| 15:29.58 | Katty | hmm |
| 15:30.00 | Katty | keeps looking |
| 15:30.23 | Naikrovek | yeah trixbox runs fine in vmware, but asterisknow can't even connect to the network at all. |
| 15:30.34 | Naikrovek | (in vmware) |
| 15:30.36 | Naikrovek | (on my system) |
| 15:30.44 | shido6 | grandpapadot: wait - you are using asterisk 1.4 on vmware workstation 6 on debain 4 or debian 5 ? |
| 15:30.51 | grandpapadot | Naikrovek: Check your vmware network settings and asterisk' bindaddr |
| 15:31.13 | grandpapadot | shido6: For development and testing, not in production, but it works pretty good. |
| 15:31.22 | shido6 | okie doke |
| 15:32.07 | grandpapadot | both 4 and 5 for two different platforms, one is an embedded system based on debian 5, the other is our core call processors based on debian 4, both asterisk 1.4 |
| 15:32.43 | Naikrovek | anyone know how to monitor the max channels open within asterisk over the past X period? |
| 15:33.19 | Naikrovek | (creating a proposal to mgmt for G.729 licenses and need to know how many I may need. |
| 15:33.21 | Naikrovek | ) |
| 15:34.06 | grandpapadot | Naikrovek: I would just look at your average calls per minute and get that many plus 33%. |
| 15:35.04 | grandpapadot | Naikrovek: Don't forget to take into account that when you switch to g729 your sound files if playned on a g729 channel and if they are not g729 will have to be transcoded. |
| 15:35.52 | Naikrovek | yeah; can they be bulk-transcoded easily? ffmpeg or something similar? |
| 15:36.02 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:36.41 | Naikrovek | i don't have enough users that lots of real-time transcoding of prerecorded sounds will be required |
| 15:38.00 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:38.23 | Katty | maybe i'll overhaul our house with zigbee products for christmas. |
| 15:39.15 | *** join/#asterisk keebler (n=Christop@adsl-75-20-234-44.dsl.rcsntx.sbcglobal.net) |
| 15:39.22 | keebler | Has anyone used the Patton M-ATA? |
| 15:39.37 | keebler | just currious of the call quality in comparison to the Linksys PAP2 |
| 15:39.57 | keebler | And power consumption, I'd like it to run off 3VDC |
| 15:40.15 | keebler | or 5vdc 500ma |
| 15:41.44 | Naikrovek | wonders if he can hack PoE into a patch panel and what kind of power supply that would require... |
| 15:41.50 | Naikrovek | and what voltage my phones are |
| 15:41.55 | Naikrovek | hrm. |
| 15:41.57 | scalex000 | hello |
| 15:42.13 | scalex000 | where to change language in asterisk with file |
| 15:42.16 | [TK]D-Fender | [11:32]<Naikrovek>anyone know how to monitor the max channels open within asterisk over the past X period? <- Go analyze your CDR |
| 15:42.21 | Naikrovek | keebler: haven't used one. |
| 15:42.38 | [TK]D-Fender | scalex000: In each channel driver config as needed or in the dialplan. |
| 15:42.46 | Naikrovek | [TK]D-Fender: tyvm |
| 15:42.54 | keebler | damn |
| 15:43.15 | scalex000 | I think in channel |
| 15:43.16 | shido6 | keeb |
| 15:43.16 | [TK]D-Fender | Naikrovek: And no, you can't jsut hack PoE in... there is a negotiation that takes place |
| 15:43.29 | scalex000 | because the voicemail speak in english |
| 15:43.30 | keebler | shido6: |
| 15:43.30 | keebler | ? |
| 15:43.40 | scalex000 | I new on this |
| 15:43.40 | shido6 | nothing - I just like saying keeb |
| 15:43.55 | Naikrovek | are you talking CDR or IEEE PoE negotiation |
| 15:44.02 | [TK]D-Fender | scalex000: Set the language in the device config for what starts the call, or set it in the dialplan. |
| 15:44.12 | [TK]D-Fender | Naikrovek: PoE <- |
| 15:44.17 | Naikrovek | k |
| 15:44.17 | keebler | Okay. Alright, if not the Patton Micro ATA, anyone try a single port FSX/ATA? |
| 15:44.27 | [TK]D-Fender | Naikrovek: CDR is for your call concurrency question. |
| 15:44.36 | Naikrovek | yes i know |
| 15:44.37 | Naikrovek | thank you |
| 15:44.42 | keebler | size is the issue |
| 15:44.49 | [TK]D-Fender | keebler: Single port generally isn't worth it. Linksys PAP2T-NA <- |
| 15:45.32 | keebler | I need to fit a WBP54G,ATA, PSU, and battery in a 4inx6inx4in box |
| 15:45.40 | keebler | linksys PAP2 is too large |
| 15:45.49 | keebler | I have 14 of the bastards |
| 15:45.57 | keebler | Only for one phone. |
| 15:46.39 | keebler | I can make the pap2 work if I increase size to 4.5in |
| 15:47.02 | [TK]D-Fender | keebler: huh? 14 ATA's for 1 phone? |
| 15:47.11 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.22 | [TK]D-Fender | keebler: What are you trying to do exactly? |
| 15:47.55 | keebler | no. 14 atas for 14 phones |
| 15:48.04 | keebler | no need for two fsx ports |
| 15:48.10 | keebler | or 5VDC/2A |
| 15:49.02 | keebler | building a new phone, and only have 14 PAP2s to chose from. |
| 15:49.37 | tzafrir_laptop | keebler, it's called FXS, not FSX |
| 15:50.09 | shido6 | FSX sounds sexier tho |
| 15:50.38 | keebler | Haha, thanks, thnking of the show |
| 15:50.41 | keebler | FX |
| 15:51.01 | TSM | get a xorcom bank or a sangoma A400 bank 12 ports per PCI slot and and a 2nd expander card |
| 15:51.03 | keebler | Anyway. FX0 FXS, regardless. |
| 15:51.18 | tzafrir_laptop | FXS is http://en.wikipedia.org/wiki/Fragile_X_syndrome , or "Martin-Bell syndrome" |
| 15:51.46 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 15:52.19 | TSM | xorcom XR0003 |
| 15:52.19 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 15:52.29 | keebler | the Patton only lists the power ass 100/240VAC external power adapter. Not what it actually consumes. |
| 15:52.31 | jpcansa | where the app Backgroud() plays the files from?? |
| 15:53.02 | hardwire | yawns |
| 15:53.11 | keebler | Damn this EEEpc. |
| 15:53.14 | jpcansa | *Background() |
| 15:53.18 | hardwire | ok so I sorta made a distribution macro. |
| 15:53.26 | hardwire | with a random strategy |
| 15:53.38 | hardwire | I need somebody to review it for sanity. |
| 15:53.57 | box2 | damn this delicious chocolate |
| 15:54.00 | box2 | DAMN IT TO HELL |
| 15:54.13 | box2 | keeps eating |
| 15:54.19 | Naikrovek | DAMNS THAT CHOCOLATE |
| 15:54.28 | hardwire | http://hardwire.pastey.net/120895 -> http://hardwire.pastey.net/120897 |
| 15:54.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:54.31 | box2 | eating cursed chocolate :( |
| 15:58.08 | *** join/#asterisk jmacz (n=mcorb@201.244.168.145) |
| 16:03.56 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 16:06.04 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:06.38 | jameswf | I just emailed and put a printed copy of http://floss.syr.edu/Presentations/oscon2006/4_How%20To%20Ask%20Questions%20The%20Smart%20Way.pdf on a co-workers dekd lmao |
| 16:08.15 | russellb | aka, "I'm an a-hole, lmao" ? |
| 16:08.21 | Kobaz | I've been seeing odd behavior on 1.6.0.10... If I have an odbc call in a hangup extension... on hangup asterisk will hit 99% cpu and kill the server for about 10 seconds, and then go back to 1% cpu |
| 16:11.20 | Kobaz | if I take out the odbc call, it's normal and doesn't max out the cpu |
| 16:12.45 | Kobaz | and the sql query that it's running is a really simple update on one column with an index... postgres doesn't even blink |
| 16:16.06 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 16:17.47 | grandpapadot | Hey guys, anyone successfully using ChanSpy() on 1.4.25+ ?? I've tried all versions from that forward and it just coredumps asterisk, there are some bug reports but not much activity. |
| 16:18.50 | grandpapadot | I'm about to try the SVN version of app_chanspy.c with 1.4.26.1 as a last resort. |
| 16:19.58 | russellb | grandpapadot: do you have a backtrace? |
| 16:20.22 | grandpapadot | No, I can generate one and open another bug report but there are some out there already ... |
| 16:22.37 | russellb | well, i'd be interested in at least a quick look at a bt on pastebin |
| 16:22.45 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 16:22.57 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:22.57 | grandpapadot | k, give me a bit |
| 16:25.11 | *** join/#asterisk woekele (i=erik@unaffiliated/woekele) |
| 16:25.38 | woekele | hey, any artdio-experts here? :) |
| 16:26.02 | woekele | I have a IPF-2002 Plus and was wondering if I could make it choose a sip realm based on the number I enter |
| 16:27.21 | woekele | right now, the only way I know of to switch to sip realm 2 and call a number is by: speaker, 2*, speaker. Then pick up phone, and dial the number. |
| 16:27.54 | woekele | but Id like to be able to dial just one number, something like: speaker, 2*<number> |
| 16:28.19 | *** join/#asterisk xpot-mobile (n=james@66-182-88-85.static.sdyl010.digis.net) |
| 16:28.45 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 16:29.31 | Kobaz | so noone has any idea why asterisk would go nuts when processing odbc in an 'h' extension? |
| 16:31.51 | zeeesh | can i add jabber.conf & gtalk.conf configuration from database ? |
| 16:32.37 | zeeesh | i want both realtime is there any feature for gtalk.conf and jabber.conf ? |
| 16:32.49 | Kobaz | i think you can add any config file from the database via static realtime and extconfig |
| 16:33.17 | Kobaz | i like static realtime versus realtime... with realtime, like with realtime sip, you lose some features |
| 16:34.49 | Naikrovek | anyone know the practical differences between a SIP trunk connecting two asterisk servers and two IAX2 trunks connecting two asterisk servers? |
| 16:35.02 | zeeesh | via static realtime what kind of statement i can enter in jabber.conf like in extensions.conf i enter "switch => Realtime"... what should be the statement for jabber.conf or gtalk.conf ? |
| 16:35.08 | Naikrovek | erm i misphrased that |
| 16:35.09 | grandpapadot | Naikrovek: Yea, there's no such thing as a sip trunk. |
| 16:35.38 | Naikrovek | well Trixbox lets me create them |
| 16:35.46 | grandpapadot | Naikrovek: SIP requires RDP (udp) and SIP (tcp) sessions for each call, iax2 configured as a trunk shares the same TCP connection. |
| 16:36.00 | Naikrovek | ah i see |
| 16:36.15 | grandpapadot | Naikrovek: Might want to ask in #trixbox, but in reality, there is no such thing as a SIP "trunk", only sip channels |
| 16:36.28 | Naikrovek | i see |
| 16:36.34 | Naikrovek | you answered my question well enough |
| 16:36.36 | grandpapadot | I find SIP more reliable. |
| 16:37.02 | *** join/#asterisk bhodder (n=blake@142.166.111.249) |
| 16:37.03 | grandpapadot | In theory, iax2 configured as a trunk should use a lot less bandwidth. |
| 16:37.22 | grandpapadot | or in reality, in my theory at least, lol |
| 16:37.59 | Naikrovek | well it would make sense that it would use a little less bandwidth |
| 16:38.12 | Naikrovek | since it's up all the time presumably |
| 16:38.24 | Kobaz | sip as a 'trunk' is much quicker for session teardown than iax is |
| 16:38.33 | bhodder | I am trying to setup my GXP 2000 with asterisk, with the GXP 2000 behind nat but it will not register. can anyone help? |
| 16:38.40 | Kobaz | an iax call will linger around for several seconds even after the call is hung up on the other end |
| 16:38.43 | TSM | grandpapadot: SIP and RDP are both UDP |
| 16:38.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:38.56 | grandpapadot | RTP |
| 16:39.09 | TSM | woops thats what i ment RTP |
| 16:39.15 | grandpapadot | RTP is udp packets, not tcp |
| 16:39.15 | TSM | both are UDP |
| 16:39.33 | grandpapadot | SIP is TCP (????) |
| 16:39.34 | TSM | you said that SIP is TCP |
| 16:39.39 | TSM | its not |
| 16:39.44 | Kobaz | there is a tcp implementation of sip |
| 16:39.47 | TSM | yes |
| 16:39.55 | Kobaz | but current implementations are all udp for the most part |
| 16:39.56 | Naikrovek | sip signalling is TCP, but the realtime data is UDP I believe |
| 16:39.57 | TSM | 1.6 and microsoft communication server |
| 16:40.05 | Kobaz | Naikrovek: both are udp |
| 16:40.16 | TSM | sip cant be tcp as i have my firewall to only allow UDP and it works |
| 16:40.16 | Naikrovek | stands corrected |
| 16:40.24 | TSM | TCP is comming in 1.6 |
| 16:40.33 | TSM | whats the point realy though |
| 16:40.41 | Kobaz | you would think it would make sense to do the signalling as tcp for the reliable delivery... but... it took people a long time to implement start implementing it |
| 16:40.51 | Naikrovek | how well do OCS and Asterisk integrate anyway |
| 16:40.56 | TSM | its a mess |
| 16:41.08 | Naikrovek | kobaz: I based my guess on that |
| 16:41.10 | TSM | they will have UDP eventualy in OCS |
| 16:41.22 | TSM | there are ways to do it |
| 16:41.43 | bhodder | I cannot get my grandstream GXP 2000 to register with asterisk and the phone is behind nat, can anyone help? |
| 16:41.43 | TSM | like have 1.6 server in between to just do the server to server calls |
| 16:41.48 | Naikrovek | I have Openfire (jabber server) working with Asterisk, but the Spark clients don't show caller ID for some reason |
| 16:41.56 | TSM | ~nat |
| 16:41.57 | infobot | from memory, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:42.05 | Kobaz | my old boss was a big udp fan... voip uses udp! so we will use udp! |
| 16:42.11 | Kobaz | it's just jeesus hell no |
| 16:42.11 | TSM | bhodder: sip_nat.conf |
| 16:42.29 | Kobaz | udp is a royal pain in the ass to program for |
| 16:42.31 | TSM | Kobaz: why does voip need TCP?.... |
| 16:42.49 | TSM | Kobaz: by the time its realised a packet has not got there its too late to send it again |
| 16:42.54 | Kobaz | TSM: SIP should be tcp... in my opinion |
| 16:43.04 | Kobaz | leave RTP as udp, that's fine |
| 16:43.05 | bhodder | do I need to config that is that the docs |
| 16:43.16 | Kobaz | but signalling shouldn't be on a connectionless, non-reliable protocol |
| 16:43.21 | Naikrovek | agreed |
| 16:43.38 | TSM | Kobaz: mabey SIP, but again not realy needed in the grand scheme of things, proof is that it works |
| 16:43.47 | Naikrovek | although, if signalling in UDP won't work, then the call won't either so maybe it's best |
| 16:43.48 | Kobaz | it does work |
| 16:43.59 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 16:44.04 | Naikrovek | would be weird to have a phone ring because TCP works but no voice because UDP gets dumped |
| 16:44.05 | TSM | bhodder: yes you need to setup your sip config correct, info at voip-info |
| 16:44.14 | Kobaz | Naikrovek: that happens plenty of times already |
| 16:44.25 | Naikrovek | really |
| 16:44.29 | bhodder | To use a phone behind nat you have to set externip and localnet |
| 16:44.35 | Naikrovek | is new to askerisk and voip in general |
| 16:44.37 | Kobaz | Naikrovek: ever work with voip with firewalls and nt? |
| 16:44.40 | Kobaz | nat |
| 16:44.48 | Naikrovek | yes |
| 16:44.54 | Kobaz | you'll hear audio from one end, but not another... or no audio at all |
| 16:45.02 | Kobaz | depending how borked the config is |
| 16:45.03 | TSM | yes ive not had any issues after i turned off the firewall doing its own SIP tansformations |
| 16:45.03 | Naikrovek | yeah firewall issue there probably |
| 16:46.11 | Naikrovek | fortunately my two asterisk sites are connected to each other via VPN and to the outside world via "sip trunk" and the firewall hasn't done anything inappropriate yet |
| 16:46.50 | Naikrovek | there are firewalls between all these locations but i guess i have them set up properly |
| 16:47.05 | Katty | http://www.tasteofhome.com/Recipes/Potato-Sausage-Casserole or http://www.tasteofhome.com/recipes/Three-Cheese-Chicken-Bake? |
| 16:47.07 | TSM | yup thats one way to do it, or use IAX2 as its all over a single UDP port and good for intersite connections |
| 16:47.18 | Kobaz | three cheese chicken |
| 16:47.34 | Naikrovek | well my two asterisk sites are connected via IAX2 and it's worked well |
| 16:47.47 | Katty | mkay. |
| 16:47.53 | Katty | makes grocery list |
| 16:48.08 | Kobaz | Katty: you need to bring some for everyone, you know |
| 16:48.10 | Naikrovek | when the SIP phones were all connected directly to one asterisk server (2/3rds of them are in India) there were all sorts of issues |
| 16:48.18 | grandpapadot | Naikrovek: What asterisk version? |
| 16:48.41 | Kobaz | Naikrovek: i've never gotten direct sip to work well |
| 16:48.53 | Naikrovek | 1.4.21.2 |
| 16:48.54 | Kobaz | Naikrovek: but a sip gateway to an asterisk box... or asterisk to asterisk... works fine |
| 16:48.57 | TSM | Naikrovek: its ok to go over VPN, but there will be a little more lag and bandwidth overhead |
| 16:49.07 | Naikrovek | i know, it works fine for us though |
| 16:49.12 | Naikrovek | 250ms or so ping times usually |
| 16:50.18 | Naikrovek | the language barrier adds a whole lot more lag to a conversation than the ping time anyway |
| 16:50.26 | TSM | haaa |
| 16:50.27 | TSM | yup |
| 16:50.41 | Naikrovek | they're great guys, just hard for me to understand sometimes |
| 16:51.32 | TSM | beat this for silly, someone in the office called their mobile from the new phone i put on their desk then told me there was a delay in the call, duh ofcourse, you have mobile on one ear and phone on the other, your brain works fast enough to tell |
| 16:51.34 | TSM | grrr |
| 16:51.36 | Naikrovek | There's one guy who I suspect thinks "hello" means "can you hear me," so until I say "yes" he keeps saying "hello" |
| 16:52.17 | Kobaz | hello! yes! |
| 16:52.27 | box2 | hahaha |
| 16:52.46 | Naikrovek | I can say "this is naikrovek" all I want but he'll keep going with the hello, until I say yes |
| 16:53.00 | Naikrovek | it's so hilarious |
| 16:53.14 | Naikrovek | and he cant' say "trixbox" it's "trixyboxy" and i laugh every time |
| 16:53.20 | TSM | i make it easy on people, they call the IT dept, i answer, 'yup' or 'wot' |
| 16:53.22 | Naikrovek | i like talking to those guys |
| 16:53.26 | Qwell | Naikrovek: that's funny. I laugh when people say "trixbox" |
| 16:53.54 | Naikrovek | yeah I need to set up asterisk on ubuntu myself and set it all up manually (again, new to asterisk here) |
| 16:53.55 | grandpapadot | lol |
| 16:54.04 | Naikrovek | can't believe how much full-cisco solutions cost |
| 16:54.54 | *** join/#asterisk RaDiC_rs (n=quassel@bchm-4d090dbc.pool.mediaWays.net) |
| 16:55.35 | Naikrovek | full cisco solution ~= $100k, Asterisk & Polycom solution ~= $15k |
| 16:55.52 | Naikrovek | I wish we'd sprung for PoE though |
| 16:55.54 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 16:56.00 | *** join/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 16:56.45 | *** part/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 16:56.48 | Kobaz | how many phones? |
| 16:56.50 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 16:56.53 | TSM | you got power for each phone then insead of PoE? |
| 16:56.53 | Kobaz | and which phones? |
| 16:57.08 | Kobaz | power to each phone works out to be around the same price as poe anyway |
| 16:57.25 | Kobaz | with polycom anyway... the power supplies are 12-15 bucks a piece |
| 16:58.14 | Kobaz | regular 24 port switch is around 200, 24 port poe switch is like $600, and it works out to be about 16 bucks a port extra, for the poe |
| 16:59.11 | Naikrovek | kobaz: that was a ballpark wild guess on price for the cisco solution but we have about 40 phones right now, most of which are polycom soundpoint 320s |
| 16:59.28 | TSM | i find the netgear switches about the right price for 26port POE |
| 16:59.29 | Naikrovek | 48 polycom power bricks are $576 |
| 16:59.51 | TSM | yep but with running poe switches its easy to put all the phones on a UPS |
| 17:00.03 | Naikrovek | ($12 per * 48) = $576 |
| 17:00.06 | Naikrovek | yeah |
| 17:00.08 | Naikrovek | that's what I want to do |
| 17:00.25 | TSM | Netgear FSM728TP |
| 17:00.35 | Naikrovek | Would like phones that I could use to page with though... |
| 17:00.48 | TSM | 24p POE Class2, and 4Gig links to cross connect |
| 17:00.57 | TSM | yup polys do that |
| 17:01.09 | Naikrovek | really? cant' figure it out on mine |
| 17:01.15 | Naikrovek | maybe i need to configure it |
| 17:01.22 | TSM | are you using freepbx? |
| 17:01.32 | Naikrovek | don't want auto-answer, but ability to use it as a pager |
| 17:01.45 | Naikrovek | yes, using trixbox which is asterisk + fop + easypbx |
| 17:02.27 | jaytee | easypbx? is that what they named the bastardized fork of freepbx? |
| 17:02.28 | TSM | its not auto-answer, but thats the way it works on the polys, it sends a SIP command with the page that makes it pick up and to handsfree then hangsup at the end |
| 17:03.22 | jaytee | TSM, that's how I do it on Polys with Alert-Info in SipAddHeader |
| 17:03.37 | TSM | yup, its all part of the freepbx prog |
| 17:03.58 | TSM | exten => s,n,Set(ALERTINFO=Alert-Info: Ring Answer) |
| 17:04.09 | Naikrovek | okay |
| 17:04.30 | TSM | either that, or get a speaker in ceiling job and page though that |
| 17:04.31 | Naikrovek | I hate relying on this freepbx to configure stuff |
| 17:04.34 | Naikrovek | going to work on that this weekend |
| 17:04.45 | Naikrovek | configuring asterisk start to finish via conf files only |
| 17:04.52 | Naikrovek | is a commandline man |
| 17:05.04 | Naikrovek | is also, ironically, a Windows guy |
| 17:05.04 | TSM | i find that once people get their heads around the custom files in freepbx, there are hooks into all scripts fairly much |
| 17:06.32 | TSM | i deal with nix boxes all day long with only cli, but for * i prefer to mainly use freepbx for most of the config, at the end of the day, there are a couple of people here that are not able to use nix and any cli, even building a computer stumps him, AND HEES IT grrrrrr |
| 17:06.37 | TSM | bang |
| 17:08.06 | *** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72) |
| 17:11.03 | IBC_Jkenney | When compiling DAHDI do i need additonal drivers for Digium Analog and Pri cards |
| 17:11.10 | IBC_Jkenney | or will DAHDI take care of it |
| 17:11.20 | jaytee | DAHDI will take care of it |
| 17:12.47 | Naikrovek | ~DAHDI |
| 17:12.48 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 17:12.48 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:13.07 | IBC_Jkenney | anyknow issues with Ubuntu 9.04 and or asterisk 1.6 out of mysql |
| 17:13.28 | Qwell | IBC_Jkenney: if there were known issues, they would be on issues.asterisk.org |
| 17:16.19 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 17:21.26 | IBC_Jkenney | Thanks for the help |
| 17:31.11 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 17:32.52 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 17:33.09 | Naikrovek | dunno who the voice of asterisk is (default menu voice) but she's annoying |
| 17:34.05 | grandpapadot | Anybody ever seen this? Just started, survived a restart, nothign has change on this switch in a while (1.4.26) http://pastebin.com/m7d1b0235 |
| 17:34.51 | Kobaz | Naikrovek: yes |
| 17:35.04 | Kobaz | Naikrovek: download the sound pack from umm, what the heck is the site |
| 17:35.15 | Kobaz | Naikrovek: voicevox or something... the girl sounds *way* better |
| 17:35.21 | Naikrovek | thanks |
| 17:35.53 | Kobaz | i forgot the site, i don't think that's it |
| 17:36.02 | Kobaz | it's something like that voicebox voxvoice voxsomething |
| 17:36.42 | *** join/#asterisk zmb (n=zm23@zaara.cuit.columbia.edu) |
| 17:37.45 | Naikrovek | switchvox? |
| 17:37.49 | Kobaz | no |
| 17:37.53 | Kobaz | they make hardware |
| 17:37.58 | Naikrovek | k just checkin' |
| 17:38.10 | Kobaz | vectorvoice voicevector |
| 17:38.28 | Kobaz | http://voicevector.com/ |
| 17:39.41 | *** join/#asterisk zmb (n=zm23@zaara.cuit.columbia.edu) |
| 17:39.51 | Naikrovek | yeah i found it same time you did |
| 17:39.55 | Kobaz | hehe |
| 17:40.02 | Kobaz | their girl is sooo much better |
| 17:40.11 | grandpapadot | Too many open files?!? BAH! |
| 17:41.46 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 17:42.34 | kaldemar | Naikrovek: allison smith - http://www.theivrvoice.com/ |
| 17:45.59 | zmb | hi all... i am using v1.4.24 and web-meetme to setup conferences... after upgrading to this version i'm seeing lots or zero bytes /tmp/ast-ami-xxxxxx files |
| 17:46.35 | zmb | only way to be able to remove these files is by shutting down asterisk and then delete them.. |
| 17:47.17 | zmb | can anyone suggest what i should check to avoid having these files? |
| 17:47.24 | zmb | left behind |
| 17:48.31 | *** join/#asterisk cb` (n=cb@72.37.252.50) |
| 17:50.46 | bhodder | I cannot get my GXP 2000 phone to work behind nat, setting the externip and localnet will not work for this, anyone have any ideas |
| 17:50.58 | Kobaz | i don't like allison smith's voice |
| 17:51.15 | Kobaz | she has a kind of "i'm better than you" tone, in the asterisk prompts especially |
| 17:51.27 | grandpapadot | Kobaz: Blasphemy! |
| 17:51.35 | grandpapadot | Kobaz: She is better. |
| 17:51.37 | Kobaz | the vector voice girl is much friendlyer |
| 17:51.40 | Kobaz | haha |
| 17:51.42 | grandpapadot | lol |
| 17:52.09 | bhodder | how can I get it to work without using a stun server? |
| 17:53.01 | kaldemar | bhodder: externip and localnet are not sufficient to get a phone to work |
| 17:53.04 | kaldemar | ~sipnat |
| 17:53.05 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:55.29 | Katty | :< |
| 17:55.44 | Katty | my company hired a new telemarketer. a kid, probably around 19 or 20. |
| 17:55.47 | Katty | complete /dork/ |
| 17:55.54 | Kobaz | is she hot? |
| 17:55.57 | Katty | male. |
| 17:55.58 | Katty | dork. |
| 17:56.00 | Kobaz | :( |
| 17:56.01 | Katty | think, table top |
| 17:56.03 | grandpapadot | Katty: Does he know how to IRC? |
| 17:56.04 | Katty | and guess what. |
| 17:56.12 | Katty | he's been following me around all friggin day |
| 17:56.16 | Katty | with this LOOK on his face3 |
| 17:56.26 | Katty | and i have seen this look. it's the OMG A FEMALE GEEK look. |
| 17:56.27 | n3hxs | Well, I can understand that Katty |
| 17:56.30 | beek | Awe? |
| 17:56.33 | grandpapadot | Katty: I'm just sayin' ... you're the one in a chat channel all day |
| 17:56.34 | grandpapadot | lol |
| 17:56.39 | beek | Infatuation? |
| 17:56.39 | [TK]D-Fender | Katty: revel in it while you can ;) |
| 17:56.41 | Katty | grandpapadot: yeah, but i'm cool person. |
| 17:56.43 | Kobaz | heh |
| 17:56.56 | Kobaz | Katty: you need some pics online. so we can share the experience |
| 17:57.02 | Katty | [TK]D-Fender: this reminds me of the first cluecon, where brc followed me around like a shadow. |
| 17:57.04 | grandpapadot | Katty: ;) |
| 17:57.07 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 17:57.16 | Katty | creepy. |
| 17:57.21 | *** part/#asterisk cb` (n=cb@72.37.252.50) |
| 17:58.03 | Katty | Kobaz: you don't need to know what i look like. pretend i'm a 50 year old guy, with a beard, who does telcom. |
| 17:58.06 | n3hxs | Tell him you are not a telephone and "go back to work" ;) |
| 17:58.11 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:58.11 | Kobaz | but but |
| 17:58.25 | Katty | no buts! |
| 17:58.31 | [TK]D-Fender | Katty: You'v just pretended to be what most people here pretend they are not ;) |
| 17:58.38 | n3hxs | knows better & has seen the pix. |
| 17:58.44 | grandpapadot | Hey! I'm only 38. |
| 17:58.48 | Katty | n3hxs: shuush! |
| 17:59.17 | n3hxs | Katty was holding an animal, but doesn't remember what kind, only Katty :) |
| 17:59.30 | *** join/#asterisk errotan (n=errotan@5403E6F6.catv.pool.telekom.hu) |
| 17:59.33 | Katty | was it a puppy or a ferret? |
| 18:00.13 | Katty | not that it matters. |
| 18:00.20 | Katty | just remember--50 year old telcom. |
| 18:01.05 | n3hxs | Black dog some brown markings, |
| 18:01.14 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:01.19 | [TK]D-Fender | Jager! |
| 18:01.23 | bhodder | Kaldemar that document still said to use the localnet and externip/ |
| 18:01.37 | [TK]D-Fender | bhodder: read the guide... |
| 18:01.39 | [TK]D-Fender | ~pb |
| 18:01.40 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:02.25 | kaldemar | bhodder: yes, but it says other things also. |
| 18:02.41 | hardwire | meh |
| 18:04.03 | Katty | [TK]D-Fender: it was probably Riddick |
| 18:04.15 | Katty | [TK]D-Fender: we adopted Jager at 4 months. |
| 18:04.21 | Katty | [TK]D-Fender: riddick at 7 weeks |
| 18:06.07 | bhodder | I have the other things set and now it registers however I get one way audio on the exten |
| 18:07.22 | grandpapadot | bhodder: do you have port 5060 tcp and ports 10000-20000 udp forwarded to your internal asterisk box as well as the localnet and externip settings? |
| 18:07.22 | kaldemar | by all means do say which way the audio goes and show your configs |
| 18:07.37 | [TK]D-Fender | bhodder: And I don't see you showing us SIP DEBUG for the failed call anywhere |
| 18:07.45 | [TK]D-Fender | bhodder: PASTEBIN is your friend. |
| 18:07.50 | bhodder | ok |
| 18:08.17 | Katty | [TK]D-Fender: pastebin is your mom. |
| 18:08.28 | kaldemar | grandpapadot: 5060 is unlikely to be a problem if he experiences one way audio. the call is already set up. and 5060 udp by default. :P |
| 18:08.36 | [TK]D-Fender | Katty: WHO'S YOUR DADDY? |
| 18:08.42 | Katty | [TK]D-Fender: i don't claim him. |
| 18:08.52 | Katty | also. |
| 18:08.53 | Katty | not you. |
| 18:09.08 | Katty | tho i don't claim you either. |
| 18:09.23 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 18:09.34 | Katty | too much sarcasm and not enough hugs. |
| 18:10.44 | Kobaz | yeah the smokey mountains |
| 18:10.45 | Kobaz | er |
| 18:12.20 | Katty | yeah, i know. |
| 18:12.23 | Katty | it's awesome. |
| 18:13.52 | bhodder | http://pastebin.ca/1528359 |
| 18:13.57 | bhodder | there it is |
| 18:14.28 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:15.06 | bhodder | what is happening now is the phone behind the nat can hear but the caller outside the nat cannot |
| 18:15.20 | Katty | rtp? |
| 18:15.34 | Katty | policy violations on the firewall? |
| 18:15.40 | Qwell | Katty: 50... ha |
| 18:15.54 | Katty | Qwell: my druid hit lvl 51 |
| 18:16.10 | Katty | Qwell: and i dual speced tank on my pally :> |
| 18:16.13 | Qwell | Katty: I was referring to your being a "50 year old guy" |
| 18:16.13 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:16.17 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 18:16.23 | [TK]D-Fender | bhodder: ........ |
| 18:16.28 | Katty | Qwell: :< |
| 18:16.33 | [TK]D-Fender | [14:07]<[TK]D-Fender>bhodder: And I don't see you showing us SIP DEBUG for the failed call anywhere <---------- |
| 18:16.43 | [TK]D-Fender | bhodder: Now please provide what we have requested. |
| 18:16.55 | [TK]D-Fender | bhodder: and you did NOT follow the guide. LOTS of settings missing. |
| 18:17.06 | bhodder | the call is not giving a fail error it goes through |
| 18:17.23 | [TK]D-Fender | bhodder: localnet=10.10.11.0 <-- BROKEN... you don't seem to understand the term NETMASK either |
| 18:17.34 | SuPrSluG | rupa:http://pastebin.com/m6c2cec9 |
| 18:17.40 | [TK]D-Fender | bhodder: Yes, and the lack of audio = BROKEN |
| 18:17.54 | [TK]D-Fender | bhodder: Now follow the guide already.... |
| 18:17.57 | grandpapadot | bhodder: It will and just send the RTP media to whatever address it thinks it needs to per your settings. |
| 18:20.44 | *** join/#asterisk bluOxigen (n=xainix20@static-host119-73-67-241.link.net.pk) |
| 18:25.56 | bluOxigen | how can i set asterisk so that people can call in to listen a long recorded mp3 files just listen, or can i do live streaming in asterisk |
| 18:26.48 | SuPrSluG | you can use ices for streaming |
| 18:27.10 | manxpower | bluOxigen: Is there a reason the recordings MUST be in mp3 format? |
| 18:27.25 | bhodder | ok I got it working now. |
| 18:27.28 | bhodder | thanks |
| 18:27.32 | SuPrSluG | and if you use a symlink it will do mp3's |
| 18:27.45 | bluOxigen | manxpower: no it could be wav also |
| 18:27.51 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:28.15 | [TK]D-Fender | bhodder: Works a lot better when you follow the instructions. |
| 18:28.21 | manxpower | bluOxigen: wav would be better from a CPU utilization and ease of setup standpoint. Just use Playback() to playback the files then. |
| 18:28.30 | bhodder | One question, if you have another phone on a different private network behind nat can you add a second localnet in general? |
| 18:28.38 | manxpower | bluOxigen: you are basically just wanting a very, very simple IVR. |
| 18:28.59 | manxpower | bhodder: localnet does not specify the REMOTE network, it specifies information about the LOCAL network. |
| 18:29.38 | manxpower | netmask specifies the address block that is local, externip/externhost is specifies the external address. anything not matching localnet will be considered external. |
| 18:29.39 | bluOxigen | manxpower: no actually my ivr is already there for a TV channel, which has 6 options in the main menu, not they want to play the either a long audio file or live streaming |
| 18:29.50 | bhodder | yes it does, I thought what I had for those in the context of that extension but i miss read that. thanks |
| 18:30.07 | manxpower | bluOxigen: there is no difference between a long audio file and a short audio file other than length. |
| 18:32.28 | [TK]D-Fender | manxpower: and maybe size ;) |
| 18:33.00 | bluOxigen | manxpower: i know - but the recording would be like an hour long, i need suggestions here, what should i propose them |
| 18:33.16 | *** part/#asterisk danfowler (n=DFowler@ip-66-9-231-226.autorev.intellispace.net) |
| 18:35.36 | manxpower | bluOxigen: There are several Playback apps. They will handle 10 second or 10 hour files. |
| 18:35.56 | bluOxigen | manxpower: thats great |
| 18:36.17 | bluOxigen | manxpower: any link wher i can get example of such IVR app |
| 18:36.49 | manxpower | exten => 6165551212,1,Playback(/path/to/really-long-wav-file) |
| 18:38.10 | manxpower | I'd use one of the control playback apps included with Asterisk then the user can rewind/fforward if they want. |
| 18:39.01 | manxpower | "core show applications like playback" |
| 18:39.23 | bluOxigen | manxpower: ok, but with this when the user will get into this option the file will be started playing from the begining, i wanted something that something similar to the onhold music tyep of thing |
| 18:40.41 | bluOxigen | manxpower: ok, that when user joing he or she might join from where where in between |
| 18:40.49 | manxpower | bluOxigen: be more specific next time. |
| 18:42.53 | *** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br) |
| 18:45.34 | bluOxigen | manxpower: ok, sorry for that, but can that be possible |
| 18:46.54 | [TK]D-Fender | bluOxigen: Script breaking the recording into segments and play them back sequentially logging what point they've reached. |
| 18:47.14 | [TK]D-Fender | bluOxigen: I'd do this in an AGI script so it sounds more seamless. |
| 18:48.59 | Naikrovek | so howcome phones use all these G.whatever codecs instead of things like mp3 or whatever? latency? |
| 18:50.02 | grandpapadot | Naikrovek: standards. Not to mention mp3 would make a terrible voice codec, the G.whatevers, some more than others, are design to work with voice specifically. |
| 18:50.26 | Naikrovek | okay |
| 18:50.45 | Naikrovek | so the big advantage of using G.729 is lower bandwidth use, yes? |
| 18:51.06 | Naikrovek | compression rather than just encoding? |
| 18:51.44 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 18:51.55 | *** join/#asterisk werdan7 (i=w7@freenode/staff/wikimedia.werdan7) |
| 18:51.59 | lowtek | Naikrovek: Even G.711 has some compression. |
| 18:52.14 | Naikrovek | yes but G.729 has significant compression, yes? |
| 18:52.23 | Naikrovek | enough to make it worth paying for, i assume |
| 18:52.23 | grandpapadot | split? |
| 18:52.50 | grandpapadot | Naikrovek: If you're using the internet for voice I think G.729 is almost mandatory for end-points that use broadband or shared connections. |
| 18:53.16 | Naikrovek | because of bandwidth use? |
| 18:53.17 | grandpapadot | Naikrovek: G.711 is approximately 84k (both directions) per active call. |
| 18:53.33 | Naikrovek | we use G.711 here right now and it's killing our bandwidth on conference calls |
| 18:53.36 | grandpapadot | Naikrovek: Wheras G.729 is around 24k or so, huge difference when you start doing a bunch of calls. |
| 18:53.38 | Naikrovek | we only have a T1 |
| 18:53.47 | Naikrovek | okay |
| 18:53.51 | Naikrovek | that's what I was looking for |
| 18:53.53 | Naikrovek | thank you |
| 18:54.13 | grandpapadot | Naikrovek: What phones are you using? |
| 18:54.21 | manxpower | Naikrovek: mp3 does not well work for pure voice. MP3 has high CPU requirements. MP3 has high latency |
| 18:54.35 | [TK]D-Fender | [14:49]<grandpapadot>Naikrovek: standards. Not to mention mp3 would make a terrible voice codec, the G.whatevers, some more than others, are design to work with voice specifically. <- Quite untrue. MP3 is lower than CD, but toll-quality is MOUNTAINS lower |
| 18:54.36 | Naikrovek | Polycom SoundPoint 320s mostly |
| 18:54.49 | *** join/#asterisk ddickenson_ (n=chatzill@67-198-0-5.static.grandenetworks.net) |
| 18:55.13 | grandpapadot | You'll have good luck with them and G.729 |
| 18:55.17 | manxpower | [TK]D-Fender: mp3 will encode lots of frequencies the PSTN cannot transport. |
| 18:55.25 | manxpower | or regular phones for that matter. |
| 18:55.37 | grandpapadot | Some phones, for example the cisco 7940, can only do a single G.729 channel at a time as the onboard processor isn't powerful enough or the software won't support it. |
| 18:55.41 | Naikrovek | I could probably set these all to G.729 at least as a first priority but if I have to conference with a phone that doesn't have g.729, I need to transcode for that user and I don't ahve a license |
| 18:56.00 | Naikrovek | I suppose I could do some testing |
| 18:56.38 | ddickenson_ | anybody tried installing fresh with dahdi-2.2.0.2? When I compile with that source from asterisk website it will not create any of the /dev/dahdi/ stuff but if I compile 2.2.0 it will. I happen to have 2.2.0 source but I'm wondering how people are doing it if they are just getting from website and if this is a known bug |
| 18:56.44 | grandpapadot | Naikrovek: Use G.729 on the desktops, G.711 for conference phones. |
| 18:57.04 | Naikrovek | why .711 for conf phones |
| 18:57.10 | [TK]D-Fender | manxpower: True, but the get ignored, voice isn't in that range therefor MP3 isn't bad for voice. PSTN is bad for NOT voice :) |
| 18:57.40 | [TK]D-Fender | And G.729 ~ 28kbps |
| 18:57.46 | grandpapadot | Naikrovek: Well you're going to free up some bandwidth switching to G.729 on your desktop phones, might as well have the wider band audio for your conference rooms. |
| 18:57.51 | grandpapadot | TK: ;) |
| 18:58.00 | manxpower | grandpapadot: if he has not license he can't convert |
| 18:58.17 | [TK]D-Fender | And you want to AvOID G.729 where at all possible |
| 18:58.55 | manxpower | Naikrovek: buy a license, it will save you time and money in the long run |
| 18:58.57 | Naikrovek | this is a 100/1000mbit network. I don't give a poop about inter-office calls, really. |
| 18:59.13 | Naikrovek | I'm planning on buying licenses, they're only $10 so it's kinda silly not to pay |
| 18:59.20 | Naikrovek | but I'm trying to work out how many I need. |
| 18:59.23 | [TK]D-Fender | Naikrovek: G.711 minimum, G.722 is four equipment supports it |
| 18:59.25 | grandpapadot | Naikrovek: You have a local asterisk server? I would run G711 locally, G.729 to an ITSP if bandwidth is an issue. |
| 18:59.46 | *** join/#asterisk luminforce (n=luminbla@cpe-70-112-179-169.austin.res.rr.com) |
| 18:59.56 | grandpapadot | Naikrovek: Then just buy as many + 33% G.729 codecs as you have outbound channels with your ITSP. |
| 18:59.58 | Naikrovek | my trunk to my VOIP provider from my * server should probably be 729, my trunk to my India office's * server should probably be 729 |
| 19:01.16 | Naikrovek | G.711 is fine intra-office |
| 19:01.34 | grandpapadot | Naikrovek: GSM is another good asterisk->ITSP codec |
| 19:01.43 | grandpapadot | Naikrovek: And no transcoding license necessary |
| 19:01.50 | Naikrovek | royalty free i'm guessing |
| 19:01.57 | Naikrovek | okay |
| 19:02.22 | grandpapadot | Naikrovek: Not sure what the general concensus is, but I think G.729 sounds more clear on average than GSM. |
| 19:02.33 | Naikrovek | okay |
| 19:03.31 | [TK]D-Fender | Naikrovek: More important that just the minimal savings from GSM (13) to G.729 (8) is the savigs of RTP overhead if you Trunk your calls to them with IAX2 saving 20kbps / call past |
| 19:03.33 | *** join/#asterisk adam0563 (n=damagele@70.103.115.194) |
| 19:03.36 | adam0563 | por fin |
| 19:03.50 | Naikrovek | k |
| 19:04.17 | [TK]D-Fender | grandpapadot: G.279 and GSM are considered damn close. GSM is better with background music, G.729 is a bit clearer on just speech |
| 19:04.47 | Naikrovek | is wondering why AsteriskNOW is only obtaining IPv6 addresses from DHCP and not v4... wtf |
| 19:05.03 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 19:05.16 | [TK]D-Fender | Naikrovek: Its just CentOS, so go dig. |
| 19:05.29 | luminforce | are these errors significant... "DEBUG[7573] chan_sip.c: = No match Their Call ID"? i'm guessing they are in relation to using DeadAGI, and the call is gone, but I'm not certain. |
| 19:05.37 | Naikrovek | yeah i know, but just reminds me why i hate linux |
| 19:05.50 | Naikrovek | why can't someone create an asterisk distro on freebsd |
| 19:06.00 | Naikrovek | soooo much nicer |
| 19:06.11 | Naikrovek | probably some kernel modules or something |
| 19:06.35 | [TK]D-Fender | Naikrovek: because there are so many fewer people to support it, DAHDI was built for Linux and BSD ports well.... lets not go there... |
| 19:06.36 | Corydon76-dig | Naikrovek: check your DHCP server to see if all IPv4 addresses are already under lease |
| 19:06.50 | [TK]D-Fender | Naikrovek: Yes, kernel modules is the issue... |
| 19:06.55 | Naikrovek | Corydon76-dig: already checked. 6-hour release |
| 19:07.15 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:07.47 | Naikrovek | a former employer had written a port of * to Java, I wish I had some of that code now. |
| 19:07.50 | Naikrovek | <-- Java programmer |
| 19:08.01 | Naikrovek | don't need DAHDI anyway |
| 19:11.35 | *** join/#asterisk sakic (n=sakic@adsl-146-134-144.clt.bellsouth.net) |
| 19:12.01 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:18.04 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 19:19.07 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 19:22.11 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 19:23.56 | *** join/#asterisk propellerhead (n=yogurt2u@200.117.115.9) |
| 19:26.53 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 19:27.55 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 19:29.09 | exsync | * to java sounds horrid |
| 19:29.22 | TSM2 | yup |
| 19:29.29 | exsync | was it for running it on solaris or something |
| 19:29.34 | exsync | the resource use had to be awful |
| 19:32.15 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 19:35.35 | Naikrovek | nah it wasn't bad |
| 19:35.44 | Naikrovek | java isnt' really bloated like it used to be |
| 19:36.20 | *** join/#asterisk Shinsaku (n=Shinsaku@chello084010157123.chello.pl) |
| 19:37.03 | Naikrovek | i've written some pretty serious middleware that i thought would be awful but turned out to perform awesomely and use relatively little memory |
| 19:37.07 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 19:37.16 | Naikrovek | high availability, high traffic stuff |
| 19:40.14 | TSM2 | cool |
| 19:40.18 | TSM2 | gona GPL it? |
| 19:41.05 | TSM2 | got a load of big projects at work, but I cant GPL them |
| 19:44.29 | *** join/#asterisk voxter (n=voxter@174.6.173.231) |
| 19:44.38 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 19:45.47 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 19:45.50 | mbrevda | http://www.securityfocus.com/archive/1/505723 <---- snom hacking |
| 19:45.50 | Naikrovek | no this was for a former employer and it's their code |
| 19:46.27 | Naikrovek | yeah. you can find a million polycom phone admin pages on google with default passwords |
| 19:48.51 | TSM2 | i need to look into the ways of securing polycom config files over wan |
| 19:50.02 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:51.34 | Naikrovek | polycom phones all have similar MAC addresses, you could maybe allow only MACs starting with 00:04:f2 to connect |
| 19:51.35 | Katty | stretches |
| 19:51.43 | lesouvage | Is there a Dutchman (or woman) on the channel with suggestion for AMD settings. We Dutch are so polite and friendly that we use to many words when we pick up the phone so AMD always detect a machine instead of human. :-( |
| 19:51.55 | TSM2 | can do that, over WAN you dont get MAC addresses |
| 19:52.08 | *** join/#asterisk zmb (n=zm23@zaara.cuit.columbia.edu) |
| 19:52.12 | shido6 | heh |
| 19:52.53 | TSM2 | MAC addresses dont traverse the local router onto the internet |
| 19:55.42 | Naikrovek | are you in control of the entire WAN |
| 19:57.29 | grandpapadot | -oO| Peace Out! |Oo- |
| 19:57.36 | TSM2 | haaa |
| 20:00.43 | bmoraca | TSM2: use FTP instead of TFTP. that why, you can verify access via credentials. |
| 20:00.53 | Naikrovek | good one |
| 20:01.03 | bmoraca | s/why/way/ |
| 20:01.47 | adam0563 | how do I set up the voice mailbox extension in Asterisk? I'm kind of new at this... I have managed one but never set one up. I know how to make users have mailboxes but not how to set up the voicemail system. Or does it come already set up? |
| 20:01.49 | Naikrovek | one could put the phones on their own subnet via dhcp, then restrict access to that subnet via iptables |
| 20:02.17 | Naikrovek | could also restrict access to web interfaces that way too |
| 20:02.56 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 20:03.20 | Naikrovek | though you could do that via password, too |
| 20:03.32 | TSM2 | bmoraca: yes i already do FTP, but everything is clear text so could be sniffed, i was thinking HTTP |
| 20:03.38 | TSM2 | woops HTTPS |
| 20:03.46 | bmoraca | that potentially works as well |
| 20:03.47 | Naikrovek | if it's a switched network it can't be sniffed |
| 20:04.05 | TSM2 | bmoraca: it works, ive done it |
| 20:04.36 | TSM2 | Naikrovek: naa not true, inline sniffer, or port mirroring will allow sniffing |
| 20:05.25 | Naikrovek | yes but that requires admin access to a switch; your regular users who would need to sniff to get it probably dont' have that |
| 20:05.39 | Naikrovek | which of your users has an inline sniffer anyway |
| 20:05.43 | TSM2 | Naikrovek: over the WAN its not under your control |
| 20:05.56 | Naikrovek | that's why i asked you earlier if you were in control of the wan |
| 20:06.09 | TSM2 | Naikrovek: inline sniffer, easy, mini PC with two ethers, done |
| 20:06.20 | Naikrovek | i'm not saying it's hard |
| 20:06.55 | TSM2 | Naikrovek: thats why HTTPS is prolly better, that way no data can be seen |
| 20:07.08 | Naikrovek | i've worked in a lot of places, some very large places, not one of them ever had a user who used an inline sniffer |
| 20:07.15 | Naikrovek | the data can be seen, it's just encrypted |
| 20:07.32 | Naikrovek | but a man in the middle attack will work with an inline sniffer too |
| 20:07.36 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 20:07.50 | TSM2 | Naikrovek: duh i know that, thats why HTTPS is better, that way no clear data can be seen, plus you can do HTTPS auth to double up |
| 20:08.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:08.13 | Naikrovek | but that's all vulnerable to your inline sniffer PC |
| 20:08.30 | Naikrovek | my point is that there is very little you can do to stop a rogue employee |
| 20:08.37 | TSM2 | Naikrovek: no its not as its all under HTTPS, you wont see whats being sent |
| 20:08.40 | Sargun | Naikrovek, layer 2 VLANs |
| 20:08.42 | adam0563 | anybody??? |
| 20:08.49 | adam0563 | how do I set up the voice mailbox extension in Asterisk? I'm kind of new at this... I have managed one but never set one up. I know how to make users have mailboxes but not how to set up the voicemail system. Or does it come already set up? |
| 20:08.53 | Naikrovek | you don't understand main in the middle |
| 20:09.16 | manxpower | adam0563: We don't support GUI installs here. |
| 20:09.20 | Sargun | Naikrovek, I mean, if you prevent someone from installing a device on the layer 2 network, doing a MITM is very hard |
| 20:09.44 | Sargun | adam0563, http://www.voip-info.org/tiki-index.php?page=Asterisk+VoiceMail |
| 20:09.52 | Naikrovek | that's true, but mac addresses can be cloned |
| 20:09.59 | manxpower | There should be a channel for whatever gui you are using |
| 20:10.32 | TSM2 | Naikrovek: i understand man in the middle, with HTTPS security certs its not possable to do MITM and see the clear data, you will only know what port and server they are connecting to |
| 20:10.32 | bmoraca | there are a hundred different ways you can prevent man-in-the-middle attacks |
| 20:10.49 | bmoraca | it's definitely possible...just very difficult |
| 20:11.10 | bmoraca | SSL is not 100%. there are vulnerabilities if you have one of the keys |
| 20:11.31 | Naikrovek | or you can impersonate the server enough to fool the phone |
| 20:11.39 | TSM2 | this is true, there is not 100% guarantee, but as far as we can go its secure as long as you are not using MD5 certs |
| 20:12.16 | raden_work | can i record like my voice menu through the phone to asterisk ? |
| 20:12.23 | Naikrovek | raden_work: yes |
| 20:12.31 | TSM2 | with the phone not seeming to support local certs then its possable to trick the phone |
| 20:13.04 | manxpower | raden_work: eventually you are going to use up all your karma by asking questions easily answered on voip-info.org and the Asterisk book. Then nobody will help you. You don't want that. |
| 20:13.04 | TSM2 | sorry it does |
| 20:13.16 | zeeesh | is there any realtime feature for jabber.conf and gtalk.conf in asterisk ? |
| 20:13.23 | raden_work | manxpower, lol thanks |
| 20:13.41 | raden_work | manxpower, this book has not helped me much |
| 20:14.25 | Naikrovek | it's stupidly easy to trick the phone. phone doesn't care if cert is valid, just that there's a cert. that cert could come from a middle man, be decrypted by that middle man, then cleartext data copied and passed to real server, then response is the same, but reverse |
| 20:14.46 | Naikrovek | you can't stop a rogue employee |
| 20:14.50 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:15.04 | Naikrovek | you can keep them honest, but you're not going to keep a sufficiently motivated person from doing anything |
| 20:16.00 | Naikrovek | not saying that you shouldn't think of security, but there's a point where extra thought and effort on the admins part does very little to add extra security |
| 20:16.35 | Naikrovek | unless the phone can be forced only to respond to valid certificates only |
| 20:16.59 | manxpower | raden_work: look in the book for the "record" application |
| 20:17.02 | TSM2 | Naikrovek: look into the docs, there are things in the phone that you can do to stop the data being seen, the phone does care about security certs, look at the docs, it even supports crypt of the config files |
| 20:17.17 | TSM2 | Naikrovek: the polys have it all |
| 20:17.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:20.02 | Naikrovek | well have fun with that. all my phones live on their own subnet and each phone has a different admin password (not the extension, not based on the extension) so if a user wants to tinker with their phone until it doesn't work then I let them, and when they break it I point them to management and they're gone |
| 20:20.39 | raden_work | manxpower, thanks |
| 20:21.21 | TSM2 | yup thats the way, im not to bothered about the phones in our office, i was talking about phones that are out of my local control but still have remote provision |
| 20:22.52 | Naikrovek | the dudes in my india office have absolutely no clue at all about how to work the phones, and they have all their passwords and all that. they used to not have any of that information but management made me give it to them. now their stuff always breaks. i take my time when fixing and always tell them to fix it first |
| 20:22.57 | Naikrovek | if they wanna F with it then they can fix it. |
| 20:23.43 | Naikrovek | i think you can even disable the web server on polycom phones on the firmware i'm using |
| 20:23.57 | Naikrovek | i should proabably do that for those dorks |
| 20:24.02 | TSM2 | power off, power on, its all sorted again, yup web server can be turned off |
| 20:24.24 | Naikrovek | no these guys have a local TFTP server and they dink with the sip.cfg file |
| 20:24.38 | Naikrovek | then they reboot all the phones to make something work and suddenly their dialplan is tweaked and i get a phone call at 2am |
| 20:24.51 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 20:24.51 | Naikrovek | i tell them to fix it and turn my phone off |
| 20:25.28 | Naikrovek | it's like giving a baby a stun gun with those guys |
| 20:25.36 | Naikrovek | they're constantly breaking their stuff |
| 20:25.37 | TSM2 | ahh, well, make all the phones provision directly back to you and keep it under your control, good thing with the phone, is as long as it gets an IP, if it cant download a config, it will just use a config from before, problem comes if they are also in control of the DHCP server |
| 20:25.40 | *** join/#asterisk af_ (n=getsmart@88-149-230-54.dynamic.ngi.it) |
| 20:25.50 | raden_work | is 133ms of latency normal on a lan ? |
| 20:25.56 | TSM2 | no |
| 20:26.11 | Naikrovek | well they know the passwords of the phones and if they don't want to change them back to me then there's nothing i can do about it. |
| 20:26.17 | Naikrovek | yeah that's way high |
| 20:26.38 | Naikrovek | in india, nepotism is a way of life |
| 20:27.13 | Naikrovek | despite being the admin i have no muscule against any of the many relatives of the office manager |
| 20:27.27 | raden_work | callcentric/17772445766 204.11.192.36 N 5080 UNREACHABLE |
| 20:27.27 | korcan | installing a new server to run out of mysql, what release do you recommend? 1.4 or 1.6? |
| 20:27.35 | korcan | I am worried about bugs in 1.6 |
| 20:27.37 | korcan | should I be? |
| 20:27.44 | Naikrovek | probably not |
| 20:27.44 | raden_work | what would be normal latency ? |
| 20:27.46 | [TK]D-Fender | checkout time, later all |
| 20:27.58 | Naikrovek | normal latency on a LAN is 0-20ms |
| 20:28.18 | TSM2 | Naikrovek: ah well, thats poo, only so much you can do before you make it too hard for you to do anything youself |
| 20:28.21 | raden_work | ive never been under 60 :( |
| 20:28.32 | Naikrovek | bad switch |
| 20:28.32 | raden_work | anyone have a idea why it'd be so high ? |
| 20:28.40 | raden_work | please say thats a joke ? |
| 20:28.43 | manxpower | qualify latency (sip show peers) is NOT network latenct |
| 20:28.45 | TSM2 | 20ms on a lan, naaa less than 5ms in a lan, depends on if you are going though routers |
| 20:28.48 | Naikrovek | a LAN? LOCAL area network? |
| 20:28.48 | manxpower | or latency |
| 20:29.06 | Naikrovek | ping time on a lan should always be < 20ms |
| 20:29.09 | raden_work | yes our local area network |
| 20:29.30 | Naikrovek | < 5 like TSM2 said if they're physically connected to the same switch |
| 20:29.38 | raden_work | ok its 0.416 |
| 20:29.47 | Naikrovek | ping time = network latency |
| 20:29.57 | Naikrovek | show sip peers is not network latency |
| 20:30.08 | *** join/#asterisk jer_ (n=jer@unaffiliated/jer) |
| 20:30.08 | raden_work | pinging why does asterisk show 101/101 192.168.1.101 D N 5060 OK (59 ms) |
| 20:30.11 | manxpower | raden_work: is that 0.416ms or 0.416s? |
| 20:30.20 | manxpower | raden_work: THAT IS NOT NETWORK LATECNCY |
| 20:30.24 | Kobaz | Naikrovek: 20ms? what is this... 1970? |
| 20:30.36 | raden_work | PING 192.168.1.101 (192.168.1.101) 56(84) bytes of data. |
| 20:30.36 | raden_work | 64 bytes from 192.168.1.101: icmp_seq=1 ttl=64 time=0.435 ms |
| 20:30.44 | Naikrovek | i think i read that the mismatch is a bug in asterisk but it could be the time it takes the phone to process the SIP message and respond |
| 20:30.51 | Naikrovek | kobaz: just wanted to give a baseline |
| 20:31.02 | Kobaz | Naikrovek: lan latency should be <1ms |
| 20:31.03 | Kobaz | heh |
| 20:31.31 | raden_work | callcentric/17772445766 204.11.192.36 N 5080 UNREACHABLE |
| 20:31.31 | Kobaz | anything more, and something is horribly wrong |
| 20:31.47 | raden_work | i did a traceroute and i can reach the desitnation i dont understand what is going on |
| 20:32.01 | Kobaz | raden_work: it means it can't reach the SIP PEER |
| 20:32.10 | Naikrovek | well i have a few machines that are right at 100 meters away and they're at <1 ms but sometimes when they're welding it screws with the network somehow and bumps it up to 5 |
| 20:32.14 | manxpower | looks to me like your have your nat config in asterisk screwed up |
| 20:32.17 | Naikrovek | cat 6 cable is going in soon |
| 20:32.27 | raden_work | 12 alpha6.callcentric.com (204.11.192.36) 44.703 ms 45.125 ms 46.387 ms |
| 20:32.28 | Kobaz | raden_work: which could mean that the port is unavailable, firewalled, unreachable, etc |
| 20:32.40 | raden_work | i just had everything up and running all last week :( |
| 20:32.40 | TSM2 | if there are old HUBs then you will get bad ping times, ive also had one machine that whenever it was turned off the ping times in the office lan would go up to 200ms, only fix was to replace the onboard NIC |
| 20:32.45 | raden_work | * pulling hair out |
| 20:32.56 | manxpower | raden_work: post your sip.conf on pastebin.ca |
| 20:33.38 | raden_work | http://pastebin.com/d5a4a9c95 |
| 20:35.02 | manxpower | raden_work: try srvlookup=no and do a sip reload |
| 20:35.22 | *** join/#asterisk ingenius (n=alektro@186.136.0.175) |
| 20:35.33 | raden_work | ok |
| 20:35.37 | zmb | hi, i'm experiencing some problem with asterisk manager interface. Using 1.4.26, when a client is connected to the manager interface, /tmp keeps filling with ast-ami-xxxxx (zero byte) files... several files are being created within a minute |
| 20:35.55 | zmb | and these files can not be deleted/removed without restarting asterisk |
| 20:36.01 | zmb | help please |
| 20:36.24 | raden_work | manxpower, why did that make all the diffrence ? |
| 20:37.29 | manxpower | raden_work: did that fix it? |
| 20:37.54 | *** join/#asterisk Michofr (n=mcho123@77.42.132.254) |
| 20:38.21 | Michofr | Hi all,When doing make install when installing asterisk on centos I got the error..error: asterisk.h: No such file or directory |
| 20:38.39 | vicTROLLA | um |
| 20:38.50 | vicTROLLA | there should probably be a cooresponding error |
| 20:39.18 | Michofr | vicTROLLA, |
| 20:39.19 | Michofr | eagi-sphinx-test.c:21:22: error: asterisk.h: No such file or directory |
| 20:39.38 | Michofr | make[1]: *** [eagi-sphinx-test.o] Error 1 |
| 20:39.57 | manxpower | Michofr: do a "df -h" and make sure you don't have any full partitions |
| 20:40.29 | Michofr | manxpower, I have 3.2 G free |
| 20:42.22 | manxpower | Michofr: something has failed terribly |
| 20:43.00 | Michofr | manxpower, How should I know what is the real issue? |
| 20:43.17 | manxpower | Michofr: my first guess would be bad memory |
| 20:43.18 | Michofr | ./configure then make and make install worked fine |
| 20:43.32 | raden_work | manxpower, yeah right away whats the deal with that |
| 20:43.48 | raden_work | how can i make 101 and 102 both ring i have exten => _X.,n,Dial(SIP/101) to make 101 ring at moment |
| 20:44.00 | Michofr | manxpower, you mean RAN or HDD memory? |
| 20:44.03 | Michofr | *RAM |
| 20:44.32 | manxpower | raden_work: srvlookup makes asterisk not connect to the host you specify. It does a DNS lookup to find the sip service for that host. Obviously callcentric screwed that up |
| 20:44.42 | manxpower | HDD does not have memory. |
| 20:44.44 | manxpower | RAM |
| 20:45.06 | raden_work | manxpower, seems like they screw everything u p |
| 20:45.37 | raden_work | manxpower, is there something they need to correct on there end ? |
| 20:46.43 | manxpower | i never turn srvloopup on |
| 20:47.07 | korcan | no opinions on 1.4 vs 1.6 with mysql backend? |
| 20:47.37 | korcan | what features would cause me problems running 1..6? |
| 20:48.19 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 20:49.00 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:49.08 | manxpower | korcan: read the UPGRADE*.txt files in the 1.6 source |
| 20:49.47 | korcan | manxpower, thanks :) I will |
| 20:51.02 | raden_work | is there a way i could call into asterisk with my cell phone have it call me back right away and dial out from asterisk ? |
| 20:51.11 | raden_work | if that made any sense at all |
| 20:51.49 | raden_work | I get free incoming on cell and I want to setup a unlimited outbound plan in asterisk |
| 20:52.31 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:52.41 | ariel_ | raden_work: yes there is a way to do that via the dial plan. You can also use DISA to give you a dial tone to dial out via the asterisk box. |
| 20:52.58 | raden_work | kewl |
| 20:53.05 | raden_work | ill look into that in future |
| 20:53.12 | raden_work | how do i make a inbound call ring all extensions ? |
| 20:53.24 | bmoraca | ~book |
| 20:53.25 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:53.35 | bmoraca | ^^^ that's how |
| 20:53.37 | raden_work | i have the book :( |
| 20:53.43 | bmoraca | did you read it? |
| 20:53.57 | korcan | is it recomended to compile from source or use the package from Ubuntu? |
| 20:55.19 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:55.22 | dupe | korcan: i vote for package, the one in ubuntu is recent |
| 20:55.43 | beek | korcan: I prefer rolling my own and compiling from source. |
| 20:55.51 | korcan | dupe, what will I NOT get using the package? Can I still install modules/etc |
| 20:55.59 | ariel_ | uses source |
| 20:56.35 | dupe | korcan: in my opinion it comes down to maintaining it for any sfotware packages like that or apache/php etc.... |
| 20:56.36 | bmoraca | monkeys are better |
| 20:56.43 | manxpower | if you use a package almost nobody will support you |
| 20:56.44 | ariel_ | the main debian is still on 1.4.12 |
| 20:56.57 | drmessano^ | 1.4.12? EWWWWWWWWWWWW |
| 20:57.08 | dupe | if its set it and forget it, package is better imho |
| 20:57.17 | drmessano^ | Not if its 1.4.12 |
| 20:57.39 | korcan | if I run 1.6 what bugs will cause me headaches? |
| 20:57.47 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 20:57.48 | drmessano^ | The bad ones |
| 20:58.00 | IBC_Jkenney | Did they ever fix the MYSQL realtime bugs in 1.6 ? |
| 20:58.07 | beek | korcan: The ones that affect functionality you need. |
| 20:58.07 | ariel_ | what bugs, it runs well, but that you will not know until you start using it... |
| 20:58.09 | IBC_Jkenney | I looked at it last summer and it made me cry |
| 20:58.28 | drmessano^ | beek: The ones that you notice? |
| 20:58.40 | IBC_Jkenney | what about realtime config from mysql |
| 20:58.43 | drmessano^ | pulls out the list of one liners for that question |
| 20:58.46 | dupe | wtf i thought ubuntu was on 1.6, and debian itself was still on 1.4 |
| 20:58.47 | dupe | damn |
| 20:59.01 | dupe | and thats why i hate debian/ubuntu in general :P |
| 20:59.03 | beek | drmessano^: the list written on that lengthly scroll? |
| 20:59.09 | drmessano^ | Yeah, Ubuntu sucks |
| 20:59.20 | korcan | how long before 1.6 is out of beta? |
| 20:59.30 | beek | korcan: It's been out of beta for a long time. |
| 20:59.34 | drmessano^ | It was out of beta last year |
| 20:59.34 | korcan | drmessano^, what is your flavor of choice? |
| 20:59.45 | drmessano^ | Do you get out much? |
| 20:59.51 | korcan | Asterisk 1.6.2.0-beta4 |
| 20:59.57 | drmessano^ | 1.6.2 BETA |
| 21:00.05 | drmessano^ | 1.6.0 NOT BETA |
| 21:00.06 | beek | The 1.6.0.x and 1.6.1.x series are production. |
| 21:00.08 | drmessano^ | 1.6.1 NOT BETA |
| 21:00.10 | drmessano^ | 1.6.2 BETA |
| 21:00.15 | korcan | ahh |
| 21:00.22 | bmoraca | lol |
| 21:00.26 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv) |
| 21:00.39 | drmessano^ | Next you'll be asking if it works with Facebook |
| 21:00.43 | bmoraca | it's like talking to a 5 year old mexican kid |
| 21:00.49 | drmessano^ | or how to Skype with it |
| 21:00.57 | IBC_Jkenney | on 1.6 in the realtime out of mysql they switched from | to , I think |
| 21:00.59 | dupe | ,,,, wait does it work with facebook now? |
| 21:01.01 | dupe | hides |
| 21:01.02 | dupe | hehe |
| 21:01.09 | aiksa[LV] | Hi everyone. Does anyone know of a method to sync ast registrations between two servers? |
| 21:01.12 | Qwell | drmessano^: ... |
| 21:01.15 | Qwell | ~skypeforasterisk |
| 21:01.16 | infobot | skypeforasterisk is probably a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://store.digium.com/productview.php?product_code=804-00019 for details on the open beta. |
| 21:01.18 | aiksa[LV] | chan_facebook? |
| 21:01.36 | IBC_Jkenney | can you use either or do you have to change all your stuff in the database |
| 21:02.30 | drmessano^ | Qwell: Just because Skype works with Asterisk doesnt mean it has any less of that douchenozzle, hello-kitty, myspace glitter cat, 16 yr old virgin feel to it |
| 21:02.55 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:03.08 | aiksa[LV] | drmessano^: it has, but it has done a fairly good job at marketing itself |
| 21:03.09 | drmessano^ | If Asterisk had a module for 4chan, would it make 4chan not full of scary emo kids? |
| 21:03.37 | drmessano^ | aiksa[LV]: Yes, and so did Hitler |
| 21:03.42 | aiksa[LV] | :) |
| 21:03.56 | drmessano^ | mein_conf? |
| 21:04.08 | aiksa[LV] | on the other hand, wo skype many even wouldnt look at the direction of voip |
| 21:04.17 | aiksa[LV] | it made it a houshold name for many |
| 21:04.34 | Naikrovek | drmessano^: :) |
| 21:04.40 | aiksa[LV] | drmessano^: LOL. voip_macht_frei |
| 21:04.53 | aiksa[LV] | not sure of the german translitteration |
| 21:05.16 | drmessano^ | If Skype didnt exist, people wouldnt think VoIP involved being tethered to a computer |
| 21:05.20 | dupe | is there a way to debug 100% cpu usage inside asterisk directly? one asterisknow system bogs down as soon as asterisk starts, doing nothing... |
| 21:05.28 | raden_work | exten => _X.,n,Dial(SIP/101&SIP/102) thats what book shows but it dont work |
| 21:05.39 | drmessano^ | dupe: Is it running 1.6? |
| 21:05.53 | dupe | nay 1.4.24 |
| 21:06.05 | raden_work | just keep getting a busy signal |
| 21:06.10 | aiksa[LV] | dupe I have seen this on a few 1.4 instalations |
| 21:06.29 | aiksa[LV] | always have thought this is due to a blocking code in my AMI integration stuff |
| 21:06.34 | dupe | it be reallly nice if * has a magicly internal cpu process debug thingie but donr think so... |
| 21:06.41 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:06.47 | aiksa[LV] | to a "probably blocking code" |
| 21:07.39 | aiksa[LV] | but back to my original question: I want to sync SIP user registrations between two asterisk servers, is it possible? |
| 21:07.51 | Kobaz | aiksa[LV]: have your sip clients register on both servers |
| 21:08.13 | aiksa[LV] | Kobaz: not all support that |
| 21:08.20 | aiksa[LV] | and there could be more than just two |
| 21:08.29 | ariel_ | why? |
| 21:08.50 | aiksa[LV] | I am setting up a concept of self replacing nodes |
| 21:09.01 | [TK]D-Fender | aiksa[LV]: use a DB for your SIP peers or run a proxy in front |
| 21:09.31 | aiksa[LV] | [TK]D-Fender: thanks. that what I was thinking off. I suppose there aint any of the shelf solutions for this |
| 21:09.44 | ariel_ | enswitch |
| 21:10.07 | [TK]D-Fender | aiksa[LV]: Run off a DB. What isn't "off the shelf" about this? |
| 21:10.31 | aiksa[LV] | [TK]D-Fender: ok, ok. :) |
| 21:11.02 | aiksa[LV] | I was thinking of doing once in a 15 secs full dump of asterisk db and import at the other end |
| 21:11.33 | aiksa[LV] | is there a way to change the sip registration holder instance? at the present it is ast db right? |
| 21:11.35 | drmessano^ | Why? |
| 21:11.39 | [TK]D-Fender | aiksa[LV]: Comparative dump. This is quite doable |
| 21:11.41 | drmessano^ | Does the DB change that often? |
| 21:12.11 | ariel_ | db replication via mysql works as well |
| 21:12.13 | aiksa[LV] | drmessano^: registrations might change |
| 21:12.19 | [TK]D-Fender | aiksa[LV]: Whats the point of this in your case? |
| 21:12.30 | aiksa[LV] | is there a way to hold registrations in mysql rather than astdb? |
| 21:13.05 | aiksa[LV] | [TK]D-Fender: I am creating my own "heartbeat service", where nodes elect master to serve the comunication. |
| 21:13.23 | aiksa[LV] | If the master should fail one of the nodes should take over Master's tasks |
| 21:13.29 | drmessano^ | Why not just use SER? |
| 21:14.00 | aiksa[LV] | drmessano^: cant. this was not my asterisk instalation, i am just one making it redundant |
| 21:14.06 | ariel_ | aiksa[LV]: see bottom http://integrics.com/products/enswitch/guides/latest/en/system/architecture/ |
| 21:14.31 | aiksa[LV] | customer himself found elastix and has cerated nightmere configuration |
| 21:15.00 | ariel_ | how many users? is this for a call center? |
| 21:15.57 | drmessano^ | HAHAHA |
| 21:16.08 | drmessano^ | Youre gonna adapt an Elastix box to this scenario? |
| 21:16.22 | aiksa[LV] | drmessano^: I am not laughing . |
| 21:16.36 | dupe | you'll be drinking soon |
| 21:16.38 | dupe | and crying |
| 21:16.44 | drmessano^ | You do realize that once you work around the stuff you wont be able to do, it will be a nightmare to even guess what you can do? |
| 21:16.49 | ariel_ | thinks he is already doing that |
| 21:17.09 | aiksa[LV] | ariel_: not large there wont be more than 50 users there |
| 21:17.12 | drmessano^ | Installer OpenSER on the boxes in front of Asterisk |
| 21:17.37 | aiksa[LV] | there are however a number of custmer made agi scripts ... |
| 21:17.39 | aiksa[LV] | :) |
| 21:18.07 | drmessano^ | That you'll need to heavily bypass the dialplan to implement |
| 21:18.27 | aiksa[LV] | drmessano^: hmm not really |
| 21:18.41 | aiksa[LV] | dialplan stays where it is |
| 21:18.50 | aiksa[LV] | the other one is "sleeping node" |
| 21:18.54 | ariel_ | wow, elastix with custom agi that the customer made... nice way to get into a mess. |
| 21:18.55 | [TK]D-Fender | ariel_: While your reg idea handles making sure both sides know where it is... what tells the PHONE where to dial? |
| 21:19.42 | drmessano^ | aiksa[LV]: I could care less about your unicorn fairyland scenario.. youre assuming youre going to integrate these scripts into the dialplan easily.. |
| 21:20.03 | kaii | hm. i've been thinking about this nut a long time know, maybe one of you can help. a lot of our customers/users accidentially transfer their callers to system functions on their busy lamp field instead of their colleage. how can i avoid this? any ideas? |
| 21:20.08 | aiksa[LV] | drmessano^: I dont have to integrate anything ... |
| 21:20.17 | drmessano^ | Sleeping node, crouching tiger, static dialplan |
| 21:20.20 | aiksa[LV] | see one box scenario is working fine now |
| 21:20.22 | ariel_ | [TK]D-Fender: we use SER for our replicated setups |
| 21:20.38 | drmessano^ | kaii: Fire the users |
| 21:20.50 | aiksa[LV] | I just need a method to transfer everything to a crouching tiger |
| 21:20.57 | kaii | drmessano^: not an option actually we service other companies |
| 21:21.22 | drmessano^ | kaii: An MSP can fire a customer, it happens all the time |
| 21:21.32 | kaii | drmessano^: i mean, can i somehow reject the transfer within asterisk? |
| 21:21.34 | TSM2 | does anyone know if the poly 330 phones advertise their power class correctly, so far the only document ive read shows that it advertises itself as a class 0 which is upto 12.95w, is it possable to force a PoE managed switch to certian classes? |
| 21:21.41 | aiksa[LV] | drmessano^: and to add to the overall mess - we are subcontractors on this. |
| 21:21.49 | TSM2 | im worried i will have 24P PoE switches and unable to use all the ports |
| 21:22.08 | drmessano^ | aiksa[LV]: backup the list of extensions and credentials, wipe the box, start over |
| 21:22.27 | drmessano^ | You'll lose money on the project before its over |
| 21:23.03 | drmessano^ | Google for "trixbox cluster" |
| 21:23.25 | ariel_ | argh trixbox yuk |
| 21:23.40 | aiksa[LV] | drmessano^: I am afraid of the result of that query |
| 21:24.23 | aiksa[LV] | "drdb raid" /me shivers |
| 21:24.29 | drmessano^ | aiksa[LV]: Yet youre using trixbox jr for this project and expect a result thats not = fail |
| 21:25.02 | aiksa[LV] | true |
| 21:25.11 | *** join/#asterisk korihor (n=korihor@190.77.83.180) |
| 21:27.33 | ariel_ | goes back to working on the new ISV setup, 3 location, load balance setup for inbound/outbound CC calling system based on Asterisk & SER |
| 21:28.59 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 21:29.04 | aiksa[LV] | ok off to sleep now. will sleep over the idea |
| 21:29.49 | aiksa[LV] | thanks for the advice, thumbs up and thumbs down as well :)) |
| 21:38.47 | *** join/#asterisk MrMeek (i=MrMeek@108-23.26-24.tampabay.res.rr.com) |
| 21:42.05 | dustybin | what is the difference between the polycom 450 and polycom 321/331 ? |
| 21:42.24 | dustybin | oh i think its the number of lines |
| 21:44.13 | dustybin | http://www.voipon.co.uk/polycom-soundpoint-ip-321-ip321-pi-1766.html |
| 21:44.54 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 21:46.23 | *** join/#asterisk hudony (n=chatzill@modemcable202.250-20-96.mc.videotron.ca) |
| 21:46.28 | hudony | hi guys |
| 21:46.49 | hudony | simple question but can't find the answer so far |
| 21:47.42 | *** join/#asterisk shinao1 (n=shinao1@41.219.210.29) |
| 21:48.10 | hudony | anyone here? |
| 21:48.52 | hudony | :( |
| 21:49.47 | *** join/#asterisk shinao1 (n=shinao1@41.219.210.29) |
| 21:50.39 | *** join/#asterisk dandate2 (n=mangy@173-11-82-122-SFBA.hfc.comcastbusiness.net) |
| 21:51.32 | lesouvage | I have some strange cdr behaviour with a call generated by a callfile and routed to a Dial() with an M() parameter in which a message is played. When hangup during first play no value of ${cdr(userfield}) is stored while I gave it a value. Any suggestion. I run 1.4.24.1 on this box. |
| 21:52.35 | lesouvage | hudony: don't ask to ask, give it a try :-) |
| 21:54.27 | *** join/#asterisk stix (n=stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 21:55.20 | stix | hi guys. I just installed dahdi and dahdi-tools. There was supposed to be a /etc/dahdi/modules.conf, but it's not there? The service is starting fine, but I'd like to disable a few modules. |
| 22:01.05 | [TK]D-Fender | dustybin: G.722, Headset jack types, screen res, # of registrations, RAM, macklight, etc.... |
| 22:01.12 | [TK]D-Fender | backlight* |
| 22:01.34 | [TK]D-Fender | stix: Probably because you didn't do "make samples" |
| 22:02.06 | stix | no, it doesn't say that in the readme - it says "make config" |
| 22:02.12 | dustybin | [TK]D-Fender: I think a 331 will be good enough for my needs |
| 22:02.23 | stix | and I got the other config-files |
| 22:02.32 | [TK]D-Fender | dustybin: Do you need the passthrough? |
| 22:04.09 | dustybin | [TK]D-Fender: I dont think so no, is the passthrough for a headset? |
| 22:05.34 | [TK]D-Fender | dustybin: No, the NETWORK switch in the 331... for plugging in-line with a PC as I mentioned before... |
| 22:05.50 | [TK]D-Fender | dustybin: If you don't need that you can save a good % and get the 321 instead |
| 22:05.54 | dustybin | [TK]D-Fender: i have a big switch |
| 22:06.02 | dustybin | right i see |
| 22:06.25 | *** join/#asterisk sjobeck (n=Adium@mail.fishsciences.net) |
| 22:07.29 | *** part/#asterisk sjobeck (n=Adium@mail.fishsciences.net) |
| 22:08.13 | [TK]D-Fender | off to martial arts & stage time. BBL |
| 22:10.08 | bmoraca | dustybin: your switch is irrelevant. whether you need the integrated ethernet switch in the phones depends on how many ports go to each desk. |
| 22:10.57 | *** join/#asterisk keebler (n=Christop@adsl-75-20-234-44.dsl.rcsntx.sbcglobal.net) |
| 22:11.35 | keebler | Is the general consensus that the PAP2T Is the best ATA for the price? |
| 22:12.19 | keebler | Anyone know how I could go about getting a custom board designed? |
| 22:12.35 | keebler | Sounds rediculous. I know. |
| 22:16.57 | keebler | I've got my PAP2 and WPB54G Running off USB right now, but it still uses too much power. |
| 22:18.29 | keebler | Well, too much for the port to really handle. the USB 2.0 Spec isn't really supposed to supply more than 500ma and 5VDC. |
| 22:19.13 | keebler | I could regulate it more, but then the PAP2 shuts down. This might not be the best channel for this. haha |
| 22:21.05 | voxter | anyone know of a bug with prepending a message to a forwarded voicemail? when i do it on this 1.4 box it just loops the prepended message over and over |
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| 22:24.27 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 22:27.44 | TSM2 | just wandering, most implimentations of poe now should be the phantom power verions over 2 pairs, so i guess its ok to use LAN splitters, what you think, having problems with not enough switches supplying full 13w per port especialy on the smaller 8 port desk switches |
| 22:28.01 | TSM2 | need to try and get most of the phones connected back to the cabinet |
| 22:29.45 | Kobaz | so frusterating |
| 22:30.03 | Kobaz | when i run odbc queries in an 'h' extension, my server grinds to a halt for 20 seconds |
| 22:30.19 | keebler | Still having that problem? |
| 22:30.21 | Kobaz | yeah |
| 22:30.28 | keebler | versionb? |
| 22:30.32 | Kobaz | 1.6.0.10 |
| 22:30.36 | Kobaz | what's crazy |
| 22:30.49 | Kobaz | is if i run the same code on a faster machine, same kernel, same asterisk, same dahdi |
| 22:30.58 | Kobaz | there is no killage |
| 22:31.07 | Kobaz | this is a 1.4ghz box |
| 22:31.20 | Kobaz | if i run it on a 2ghz dual core.. asterisk stays at like 1-5% cpu |
| 22:31.20 | keebler | Was going to say, never experienced the problem on my Atom proc machine. |
| 22:31.31 | Kobaz | if i run it on a 500mhz box, the box will completely lock up |
| 22:31.54 | keebler | OS? |
| 22:32.00 | Kobaz | I'll have to compile asterisk with debugging symbols and run it through a profiler |
| 22:32.05 | Kobaz | 2.6.23.9 |
| 22:32.13 | keebler | Distro? I'm running it on FBSD. |
| 22:32.25 | Kobaz | linux |
| 22:32.31 | Kobaz | i use debian, but i compile my own kernels |
| 22:33.24 | keebler | hmm. So its a 1.4ghz server? And performance is dependant on processor speed? |
| 22:33.39 | Kobaz | yeah |
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| 22:34.27 | Kobaz | lemme try on my laptop |
| 22:34.27 | TSM2 | what poe switches do people use? what split loads do you run |
| 22:34.39 | Kobaz | TSM2: linksys/netgear |
| 22:35.28 | TSM2 | Kobaz: have you got poly phones? |
| 22:35.50 | keebler | Anyone in the DFW/Texas area? |
| 22:35.59 | Kobaz | TSM2: yeah |
| 22:36.11 | Kobaz | keebler: things will go REALLY slow if there are any database errors |
| 22:36.16 | Kobaz | keebler: twice as slow |
| 22:36.44 | keebler | Kobaz: Hmph. Going over the docs. |
| 22:37.43 | TSM2 | TSM2: problem im seeing is that Poly 330 do not advertise themselfs as Class2 units, they show up as Class0 so the switches will allocate the wrong power to them and 24p netgear switch cant handle more than 14/15 phones per switch, how have you found them, mabey they changed somthing in firmware but did not put it in the changelog |
| 22:38.06 | TSM2 | Kobaz: that was ment to be you, not me :) |
| 22:38.51 | keebler | hah |
| 22:40.19 | Kobaz | keebler: my laptop too it's a 1.6ghz |
| 22:40.25 | Kobaz | keebler: same asterisk versions and etc |
| 22:40.35 | keebler | I'm running 1.4 |
| 22:40.47 | keebler | Shouoldnt make a difference though. |
| 22:41.08 | Kobaz | well |
| 22:41.09 | Kobaz | it does |
| 22:41.15 | Kobaz | lots of code changes between 1.4 and 1.6 |
| 22:41.39 | Kobaz | tsm2: yeah i've seen that before |
| 22:42.01 | Kobaz | tsm2: mostly i have 24 port switches that had 12 port poe, and everything is okay |
| 22:43.39 | TSM2 | Kobaz: thas poo, well with that scaling i would need 3 switches then, not gona happen, i found that 3COM 4210 PWR switches have full 13w for all 26ports |
| 22:43.59 | Kobaz | keebler: so, it's actually not specifically odbc |
| 22:44.24 | Kobaz | keebler: it has to do with the 'h' extension in general... if i do say... 5 TrySystem()'s in a row, of something innocuous like cat hi > /tmp/bob |
| 22:44.40 | Kobaz | keebler: then the machine will be brought to it's knees util it finishes |
| 22:45.06 | TSM2 | Kobaz: each netgear FS728TS switch is 300GBP while the 3coms are 450GBP |
| 22:45.19 | hardwire | when will g729 for 1.6.2 come out? |
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| 23:19.24 | manxpower | Read the docs! Read the docs! |
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| 23:34.28 | Kobaz | why do sip peers show up as 'Unmonitored' if I use static-realtime |
| 23:36.08 | Kobaz | well I'm really happy I have a fix for it now... (not using -p) |
| 23:36.09 | Kobaz | er |
| 23:36.15 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 23:36.55 | Kobaz | anyways... iax2 peers statuses get updated, using static-realtime, but not sip |
| 23:37.56 | Kobaz | k |
| 23:37.58 | Kobaz | er |
| 23:38.08 | manxpower | Kobaz: "Unmonitored" normally means "qualify=" is not set or is set to "no" |
| 23:38.16 | Kobaz | oh hmm |
| 23:38.20 | Kobaz | maybe i forgot to set that |
| 23:38.30 | manxpower | looks over his glasses at Kobaz |
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