| 00:05.00 | *** join/#asterisk dvsensey (n=rraz@196.207.243.230) |
| 00:05.17 | dvsensey | hi all |
| 00:13.09 | *** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr) |
| 00:17.58 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 00:27.27 | *** join/#asterisk joesuffceren2 (n=chatzill@srv.fgp.com) |
| 00:28.41 | joesuffceren2 | I need some help setting up asterisk to log to MSSQL via freetds. I've installed freetds and recompiled asterisk with freetds support. This pastebin has my freetds.conf file and cdr_tds.conf file pasted in: http://pastebin.com/m61e9aea4 |
| 00:30.03 | joesuffceren2 | When I start asterisk, I was getting an error saying "ERROR[6880] cdr_tds.c: Failed to connect to MSSQL server." but after correcting my server name, I am now getting "VERBOSE[7010] logger.c: == Parsing '/etc/asterisk/cdr_tds.conf': [Jan 2 19:22:28] VERBOSE[7010] logger.c: Found" which seems like a success message, but I get no output logged to cdr when I place calls |
| 00:30.19 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 00:31.50 | [TK]D-Fender | joesuffceren2: Just because if can read your config file doesn't mean it can ACT on it |
| 00:32.15 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:32.41 | joesuffceren2 | ok. I thought that the absence of an error message was significant, though? |
| 00:33.15 | [TK]D-Fender | joesuffceren2: Go dump all related configs including where you specifiy to use your DB for this |
| 00:34.07 | joesuffceren2 | dump as in remove them from the asterisk box or dump as in put them in a pastebin for you? |
| 00:35.20 | joesuffceren2 | sorry if that's a dense question, but I'm not quite following you |
| 00:38.03 | [TK]D-Fender | joesuffceren2: PB |
| 00:49.02 | joesuffceren2 | sorry for the delay. The only configs that I know about are the freetds.conf and the cdr_tds.conf. both of those are in the pastebin I posted above. What other configs would you like? |
| 00:49.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:51.07 | mchou | [TK]D-Fender: 3 way calling in asterisk works even w/o meetme, coorect? |
| 00:52.02 | [TK]D-Fender | mchou: I'm uncertain for Zap FXS, but for all others its handled by the device, not * |
| 00:52.34 | mchou | [TK]D-Fender: ok, thanls |
| 00:52.37 | mchou | thanks* |
| 00:53.39 | *** join/#asterisk fashnek (n=andrewfo@64.201.247.2) |
| 00:55.19 | fashnek | can anyone with more asterisk experience than me just quickly assess this scenario and tell me if it's possible with asterisk? |
| 00:55.49 | fashnek | I really don't want to chase something that can't be done here |
| 00:56.29 | jaytee | ~ask |
| 00:56.30 | jbot | rumour has it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:57.05 | [TK]D-Fender | joesuffceren2: I do not see you specifying a table in your TDS config |
| 00:57.06 | fashnek | I want to be able to make multiple outbound that all "feed" to my extension -- i.e., I hear everything occurring on all calls, and all calls hear me -- but they do not hear each other |
| 00:57.14 | fashnek | outbound calls, that is |
| 00:58.36 | joesuffceren2 | [TK]D-Fender: I didn't specify one since I used the table name cdr and that's what asterisk assumes. Should I specify one anyway? When asterisk starts it says "no table name specified. Assuming cdr" |
| 00:58.58 | [TK]D-Fender | joesuffceren2: Beyond that I'm not sure what to advise... |
| 00:59.14 | [TK]D-Fender | joesuffceren2: Do you see any error messages on the DB side? |
| 00:59.37 | mchou | fashnek: how is that different than intercom? |
| 01:00.10 | [TK]D-Fender | mchou: bi-directional (limited) |
| 01:00.12 | mchou | fashnek: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom |
| 01:00.20 | mchou | #3 seems to apply |
| 01:00.22 | [TK]D-Fender | fashnek: No... I don't see any way of doing this. |
| 01:00.25 | fashnek | I am completely unfamiliar with intercom; can is be bidirectional? |
| 01:01.19 | [TK]D-Fender | fashnek: Non-applicable. |
| 01:03.33 | joesuffceren2 | [TK]D-Fender: I don't see any errors in the SQL logs. Do you know of any working configs posted anywhere that I could take a look at? there are examples in the default .conf files, but those were what I followed to get where I am... FWIW, there wasn't a table name specified in the example .conf file either |
| 01:04.09 | joesuffceren2 | scratch that, there is a table specified in the default (sorry) trying that now |
| 01:04.42 | [TK]D-Fender | joesuffceren2: My sample had "table=cdr", but ok... no I don't have any specific resources for this.... I'd suggest you Google for a while. |
| 01:06.20 | fashnek | [TK]D-Fender: Why would a duplex Page() not be appropriate? |
| 01:08.04 | *** join/#asterisk Supergrilo (n=fabio@unaffiliated/lovezinho) |
| 01:08.12 | *** part/#asterisk Supergrilo (n=fabio@unaffiliated/lovezinho) |
| 01:08.13 | [TK]D-Fender | fashnek: Go try, because this uses Meetme and I doubt it'll isolate the ends |
| 01:08.18 | [TK]D-Fender | fashnek: at BEST |
| 01:08.47 | joesuffceren2 | the sample cdr.conf recommends that you use [global] in cdr_tds.conf. When I do that, I get "could not connect to MSSQL server." When I change [global] to [SQL-TEMP] (my db name as specified in freetds.conf) I don't get the error message, but maybe that's because it's not trying |
| 01:08.53 | fashnek | and by failing to isolate the ends you mean that they will hear each other? |
| 01:09.40 | joesuffceren2 | specifying a table did not help. :-( I've already done googling, and the info is pretty sparse, but I'll do some more. Thanks for the help. I'll keep this window open. Feel free to throw a suggestion my way if you think of any |
| 01:10.43 | fiXXXerMet | OK, still having intermittent audio problems. Ports are setup, and my sip_nat.conf and sip.conf seems to be setup correctly as well. I have also noticed that sometimes when the other caller hangs up the phone, asterisk doesn't register the hang up at all and keeps my line open |
| 01:12.06 | [TK]D-Fender | fashnek: Yes, I would call that "failing" |
| 01:12.24 | [TK]D-Fender | fiXXXerMet: And you know there is no way I'd take this on faith, right? |
| 01:12.30 | [TK]D-Fender | fiXXXerMet: PASTEBIN |
| 01:12.33 | fiXXXerMet | I am working on it |
| 01:21.04 | fiXXXerMet | [TK]D-Fender: http://pastebin.com/m762392f2 |
| 01:22.26 | [TK]D-Fender | fiXXXerMet: externip=public.ip <- thishas an actual IP in your real config? |
| 01:22.31 | fiXXXerMet | yes |
| 01:23.15 | [TK]D-Fender | fiXXXerMet: then if the numbers are right [general] and that 1 peer seem OK. |
| 01:23.22 | [TK]D-Fender | fiXXXerMet: what do you have forwarded to *? |
| 01:24.05 | fiXXXerMet | [TK]D-Fender: forwarded to *? |
| 01:24.20 | fiXXXerMet | oh duh, sec |
| 01:27.26 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 01:28.04 | fiXXXerMet | Probably more than I need, as I have been adding things as I have read other articles, but http://pastebin.com/m7279f4e2 |
| 01:28.19 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 01:28.32 | fiXXXerMet | That is tcp/udp | from | port | to | n/a | status |
| 01:31.19 | [TK]D-Fender | 10001 should be 10000 |
| 01:31.33 | [TK]D-Fender | fiXXXerMet: the rest loooks fine. |
| 01:31.47 | fiXXXerMet | rtpstart is set to 10001 |
| 01:31.52 | [TK]D-Fender | fiXXXerMet: make sure all of your phones are set to "canreinvite=no" |
| 01:32.03 | [TK]D-Fender | fiXXXerMet: ok, that should cover that then. |
| 01:32.20 | fiXXXerMet | Each extension is set to canreinvite=no |
| 01:32.22 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:32.22 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:32.56 | *** part/#asterisk tictac (n=tictac@c-24-18-95-252.hsd1.wa.comcast.net) |
| 01:33.29 | sah-work | strange question. i have an aastra 480i and cannot seem to turn off the backlight |
| 01:33.51 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 01:34.14 | sah-work | anyone have any thoughts? |
| 01:34.29 | [TK]D-Fender | sah-work: What does its admin guide say? |
| 01:36.39 | *** join/#asterisk neoark (i=na1du@unaffiliated/mukhbiir) |
| 01:36.57 | *** part/#asterisk neoark (i=na1du@unaffiliated/mukhbiir) |
| 01:38.25 | [TK]D-Fender | sah-work: mine seems to say "nothing". So unless its accessible on the phone's local interface the answer appears to be "impossible" |
| 01:38.37 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 01:38.41 | eppigy | hello |
| 01:38.42 | eppigy | i am dave |
| 01:38.58 | justdave | so am I |
| 01:39.09 | eppigy | yes |
| 01:40.26 | [TK]D-Fender | these are both factual |
| 01:40.32 | sah-work | okay. that i want i wanted to know. seems strange. not used to dealing with these. i did check the docs. hopped i missed something |
| 01:40.54 | fiXXXerMet | [TK]D-Fender: any other ideas? |
| 01:41.07 | [TK]D-Fender | fiXXXerMet: Get real SIP debug from a real call. |
| 01:41.14 | [TK]D-Fender | (that failed) |
| 01:41.27 | fiXXXerMet | ok |
| 01:42.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:43.49 | fiXXXerMet | [TK]D-Fender: There is a lot of output - http://pastebin.com/d2ddb695c. In this call, neither of us could hear each other |
| 01:45.44 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-5e2ecb1510920120) |
| 01:46.48 | [TK]D-Fender | fiXXXerMet: do sip reload.... |
| 01:47.15 | [TK]D-Fender | fiXXXerMet: Peer audio RTP is at port 192.168.9.17:5052 <- doesn't look right |
| 01:47.48 | [TK]D-Fender | fiXXXerMet: -- Executing [s@macro-dialout-trunk:20] Dial("SIP/5003-095c0040", "SIP/Vitelity-Outbound/4103703252|300|") in new stack |
| 01:47.58 | [TK]D-Fender | fiXXXerMet: Please dump this peer masking only PW |
| 01:48.11 | eppigy | YOU CAN LEAVE THE PW FOR ME |
| 01:48.16 | eppigy | whoops |
| 01:48.35 | jaytee | yeah, cuz we all trust "dave" implicitly |
| 01:48.43 | eppigy | as well you should |
| 01:49.04 | eppigy | i am like family |
| 01:49.19 | fashnek | I trust dave explicitly |
| 01:50.46 | eppigy | i like where this is going |
| 01:51.13 | fiXXXerMet | [TK]D-Fender: Any easy way to see only debug output for that peer? |
| 01:51.27 | fashnek | grep? |
| 01:52.22 | [TK]D-Fender | fiXXXerMet: do a SIP reload and try your call again. We see it not counting that peer as NAT'd |
| 01:52.50 | fiXXXerMet | [TK]D-Fender: ATM I am using a softphone from home (on a different lan) |
| 01:52.52 | [TK]D-Fender | fiXXXerMet: So while the config looked OK, I'm thinking its not in effect |
| 01:53.02 | fiXXXerMet | would that cause it? |
| 01:53.08 | [TK]D-Fender | fiXXXerMet: Unapplied changes |
| 01:53.14 | fiXXXerMet | ok |
| 01:54.35 | [TK]D-Fender | fiXXXerMet: Everything else checks out, though for what I asked earlier just makesure your vitelity entries are "nat=no" |
| 01:55.18 | fiXXXerMet | they didn't have that |
| 01:55.32 | fiXXXerMet | let me try again |
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| 02:21.23 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 03:00.06 | dug | I am trying to determine if asterisk can see my zaptel interface, I have a digium card and I had the modules working but I screwed up my install files.. I have my interfaces in zapata.conf but when I run rasterisk -rvvv I dont see the interface ring |
| 03:00.35 | dug | when I run ztscan I see the interfaces |
| 03:01.06 | dug | same with ztcfg -vvv |
| 03:03.23 | [TK]D-Fender | dug: pastebin your zaptel & zapata.conf, "zap show channels" from * CLI |
| 03:08.49 | *** join/#asterisk LeO_22 (n=LeO@157-128-22-190.adsl.terra.cl) |
| 03:08.55 | LeO_22 | hi! |
| 03:10.12 | drmessano | HELLO!!! |
| 03:10.18 | LeO_22 | i need some help !! |
| 03:10.20 | LeO_22 | :P |
| 03:10.24 | drmessano | OF COURSE !! |
| 03:10.41 | LeO_22 | do you know how to setup ldap in asterisk :P??? |
| 03:11.00 | LeO_22 | (I'm from chile.. my english is not so good :P) |
| 03:11.35 | drmessano | I'm from the U.S., my ldap is not so good. SORRY!! |
| 03:11.50 | LeO_22 | buuu.. |
| 03:11.59 | LeO_22 | well |
| 03:12.06 | LeO_22 | thanks anyway |
| 03:13.25 | LeO_22 | do you know how to make work ... asterisk and zimbra cs, like one unike system? |
| 03:13.59 | dug | [TK]D-Fender: http://pastebin.com/m6326e92e |
| 03:14.58 | [TK]D-Fender | LeO_22: you are making no sense at all. What wil LDAP have to do with *? |
| 03:15.28 | LeO_22 | zimbra works with ldap.. |
| 03:15.38 | LeO_22 | i know asterisk do that too.. |
| 03:15.47 | drmessano | What do you want them to talk about? |
| 03:15.53 | drmessano | What is the point of the exchange? |
| 03:16.06 | LeO_22 | the thing is.. i want to use just one ldap directory to the two system.. |
| 03:16.47 | LeO_22 | then.. when i create an user in ldap directory... this user will exists in asterisk and zimbra... |
| 03:16.55 | LeO_22 | it this possible?. |
| 03:17.08 | [TK]D-Fender | LeO_22: Asterisk is not some mailing list or e-mail server. It has nothing in common with these |
| 03:17.35 | [TK]D-Fender | LeO_22: Perhaps if you created extra field they may have more in common. go read THE BOOK |
| 03:17.37 | [TK]D-Fender | ~book |
| 03:17.37 | jbot | extra, extra, read all about it, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 03:17.39 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 03:17.46 | dug | [TK]D-Fender: looks like asterisk didnt build with zaptel... rebuilding |
| 03:17.54 | [TK]D-Fender | dug: "zap show channels" <---------- |
| 03:17.54 | jaytee | it's only a matter of time before TrixboxZ is released. |
| 03:18.18 | [TK]D-Fender | jaytee: "wit a Z y0!" |
| 03:18.19 | dug | [TK]D-Fender: no zap command... |
| 03:19.10 | jaytee | "Now with TrixboxZ you can screwup your VoIP phone system and your email server all through one ugly GUI." |
| 03:19.21 | jaytee | dug, what version? |
| 03:19.24 | LeO_22 | but zimbra can work with a fone number.. i need waht that number to be an asterisk number...(i'm a newbie in asterisk :$) |
| 03:19.40 | dug | jaytee: 1.4.22 |
| 03:19.45 | jaytee | thought so |
| 03:20.22 | jaytee | [TK]D-Fender, I had the same problem. chan_zap no longer appears when you go into make menuselect. I think it forces you to use DAHDI. |
| 03:20.45 | jaytee | unless there's some poorly documented edit of the make file |
| 03:21.08 | LeO_22 | trixbox come with ldap??? |
| 03:21.29 | jaytee | LeO_22, I was making a joke |
| 03:21.37 | [TK]D-Fender | dug: LeO_22 If you're an * newb, then your request is like asking how to FLY berfore learn how to CRAWL |
| 03:21.53 | LeO_22 | :P |
| 03:21.55 | jaytee | LeO, the only thing Trixbox comes with is.......PAIN!!! |
| 03:22.05 | [TK]D-Fender | LeO_22: go read the book. Iyou are in way over your head |
| 03:22.13 | LeO_22 | ok... |
| 03:22.15 | LeO_22 | thanks... |
| 03:22.29 | [TK]D-Fender | LeO_22: Its free, use it.. |
| 03:22.40 | jaytee | russellb you awake? |
| 03:22.54 | dug | [TK]D-Fender: I am not an * newbie |
| 03:23.12 | russellb | depends why you're asking :) |
| 03:23.40 | [TK]D-Fender | dug: good, I'll rate you as a "casual strol" trying to "jog" :) |
| 03:24.00 | LeO_22 | i'm a little desperate |
| 03:24.31 | [TK]D-Fender | LeO_22: Sorry, nobody is going to try to carry you though a complete setup of this.... |
| 03:24.35 | jaytee | russellb, I'm asking about zaptel support with 1.4.22 It doesn't show up anymore when you run make menuselect. Is this by design? I thought 1.4.22 would support either zaptel or dahdi? |
| 03:24.48 | dug | [TK]D-Fender: lets go with casual stroll but I fixed my issue ... it was a build issue |
| 03:25.00 | russellb | jaytee: it's supposed to support both zaptel and dahdi, maybe it was a bug ... |
| 03:25.06 | [TK]D-Fender | dug: See, only a small jump :) |
| 03:25.11 | LeO_22 | ofcourse.. i just looking for a guide.. some help... |
| 03:25.17 | [TK]D-Fender | ~book |
| 03:25.18 | jbot | it has been said that book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 03:25.21 | [TK]D-Fender | GUID ^^^^^^^^^^^^^^ |
| 03:25.39 | russellb | the book says nothing about using the asterisk support for ldap |
| 03:25.45 | russellb | it's so new that I doubt any documentation exists for it |
| 03:25.48 | [TK]D-Fender | LeO_22: "some help" applies when you are "somewhere". Right now you seem to be "nowhere" |
| 03:25.52 | jaytee | russellb, must be some kind of bug with CentOS 5.2 because several people have been in here this week with the exact same problem. |
| 03:26.05 | [TK]D-Fender | 1 earlier today |
| 03:26.07 | russellb | jaytee: *shrugs* |
| 03:26.13 | drmessano | CentOS 5.2 is NOT the problem.. nope |
| 03:26.21 | drmessano | kills jaytee and hides the body |
| 03:26.26 | drmessano | Yep, see.. not a problem |
| 03:26.55 | jaytee | drmessano, didn't think it was. I think it's something with the very latest zaptel and 1.4.22 Haven't tried with the next earlier release of Zaptel though |
| 03:27.34 | drmessano | Yeah, I dunno.. I jumped on 1.6 as fast as I could. 1.4 was getting too stable and I was hoping for something more to complain about |
| 03:27.43 | jaytee | lol |
| 03:28.01 | drmessano | Guess I shudda jumped on the alphas |
| 03:28.07 | drmessano | or even an early beta |
| 03:28.18 | LeO_22 | i've got asterisk working.. |
| 03:28.30 | LeO_22 | i was looking for ldap support.. |
| 03:28.39 | drmessano | My latest tin foil hat theory: 1.6 is 1.4 with the version numbers SED'ed out. |
| 03:28.42 | LeO_22 | y got asterisk-ldap plugin |
| 03:28.44 | drmessano | Not "really" 1.6 |
| 03:29.14 | jaytee | I'm running 1.6.0 with DAHDI here at home but I don't have time to debug any issues that might crop up moving to 1.6 on my IVR server. Especially with Lumenvox. |
| 03:30.01 | LeO_22 | zimbra works with ldap directory.. |
| 03:30.03 | drmessano | Sad part is, I think some devs on 3rd party apps that are only "somewhat serious" about Asterisk are gonna be a while jumping on 1.6 due to some notion that the uptake will be slow |
| 03:30.20 | *** part/#asterisk fiXXXerMet (n=kyle@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 03:30.32 | drmessano | I've noted a couple apps where the devs even claimed the apps worked with 1.6 and hadnt even TESTED |
| 03:30.42 | drmessano | kicks fly-by-nighters |
| 03:30.55 | bkw_ | drmessano: how are you these days? |
| 03:30.56 | LeO_22 | but i don't know how to make talk both ldap directories.. |
| 03:31.09 | drmessano | bkw_: Fantastic. You? |
| 03:31.12 | bkw_ | great |
| 03:31.28 | bkw_ | loving my siren7 and siren14 codecs |
| 03:32.51 | jaytee | drmessano, I did an install today of CentOS 5.2 on a Dell PowerEdge 1750. I then installed the current tarball of zaptel and asterisk 1.4.22 chan_zap did not show up in make menuselect. I scrubbed and started over and used earlier builds of zaptel and * 1.4.21 and it all works like a champ. |
| 03:33.07 | drmessano | Hmmm |
| 03:33.20 | bkw_ | luves 32kHz voip |
| 03:34.01 | jaytee | bkw_, what phones are you using with it? Polycom with HD? |
| 03:34.06 | drmessano | I've looked at the G722 stuff.. Need to wait for the handsets to come down some |
| 03:34.19 | bkw_ | G722.1C and G722.1 with my ip 6000's |
| 03:34.29 | bkw_ | jaytee: I wrote the G722.1C and G722.1 module for FreeSWITCH |
| 03:34.52 | jaytee | wow! |
| 03:35.01 | jaytee | wish I could code like that |
| 03:35.14 | jaytee | I can barely get past "hello world!" |
| 03:35.21 | bkw_ | I didn't write the lib for the codec... but Steve Underwood and I worked together on that one |
| 03:35.24 | drmessano | had himself at "hello" |
| 03:35.33 | jaytee | lol |
| 03:35.58 | bkw_ | jaytee: I also wrote the celt codec module in FreeSWITCH for 48kHz voip |
| 03:36.22 | jaytee | bkw_, go forth and brag no more!!! |
| 03:36.36 | drmessano | I had been looking at G722 for replacing older, expensive, BRI ISDN based remote broadcast setups |
| 03:36.45 | bkw_ | G722 can run over the PSTN |
| 03:37.10 | LeO_22 | any help???... <[TK]D-Fender> don't get furious : |
| 03:37.13 | drmessano | G722 is a bit of a standard in that relm |
| 03:37.22 | bkw_ | yah if both ends can support the capability |
| 03:38.15 | [TK]D-Fender | LeO_22: You aren't asking for help, you're asking for someone to do it for you and you've been pointed towards docs on this. |
| 03:38.25 | LeO_22 | someone could explain me how works ldap?? :D |
| 03:39.07 | drmessano | I know the Zephyr ISDN codecs use G722 extensively in many remote broadcast setups, and the quality over 128k ISDN is phenominal |
| 03:39.25 | drmessano | Actually sounds better than most of the studio gear used to record/mix it |
| 03:39.26 | [TK]D-Fender | LeO_22: http://www.google.ca/search?hl=en&q=asterisk+ldap+integration&btnG=Google+Search&meta= |
| 03:39.29 | coppice | G.722 only uses 64k |
| 03:39.56 | [TK]D-Fender | coppice: kbit? |
| 03:40.02 | drmessano | For stereo, in this case |
| 03:40.03 | coppice | yes |
| 03:40.08 | drmessano | "broadcast: |
| 03:40.10 | drmessano | "broadcast" |
| 03:40.15 | LeO_22 | thanks... |
| 03:40.22 | coppice | ah. G.722 is about AM radio quality |
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| 03:40.44 | [TK]D-Fender | coppice: thats the same rate as G.711 though. |
| 03:40.51 | coppice | yes |
| 03:40.54 | jaytee | bkw_, how are you getting 32kHz with an IP 6000? it says on Polycom's page it handles up to 14kHz. |
| 03:41.00 | [TK]D-Fender | coppice: How does that work? |
| 03:41.27 | coppice | kinda like G.729 is only 8k :-\ |
| 03:41.37 | drmessano | Better codec |
| 03:41.43 | drmessano | Size doesnt matter |
| 03:41.52 | drmessano | Does more in the same space |
| 03:42.30 | coppice | G.722 is not that great a codec. it was designed to be reasonable cost in 1985 silicon, rather than super duper |
| 03:42.33 | [TK]D-Fender | coppice: yes, G.729 is 8K because its a heck of a lot of lossy compression compared to say G.711. How can G722 be so much better than G711 in the same BW? |
| 03:42.58 | coppice | So what? G.711 is also lossy compression. |
| 03:43.26 | [TK]D-Fender | coppice: companded I get, but nothing extra funky... |
| 03:43.48 | [TK]D-Fender | coppice: all very flat-rate isn't it? |
| 03:43.59 | coppice | its still lossy compression. lossy compression is not something bad. its something good, if you do it right |
| 03:44.01 | bkw_ | jaytee: go read up on audio a bit more then you'll know |
| 03:44.11 | drmessano | It would be fascinating to have an 2-port ATA supporting G722. Could use simple POTS mixers to create the same quality as a $4000 ISDN box |
| 03:44.20 | bkw_ | coppice: wrote the g722.1/c lib I used in FreeSWITCH |
| 03:44.24 | bkw_ | ;) |
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| 03:45.00 | [TK]D-Fender | drmessano: And what about the 2-port ATA that actually supports even G.729 on both channels? :) |
| 03:45.14 | drmessano | SPA-2102 will |
| 03:45.14 | [TK]D-Fender | drmessano: Walk before you run! |
| 03:45.23 | coppice | Think about it. Any bit rate reduction you do in a constant bit rate channel is lossy. |
| 03:45.23 | [TK]D-Fender | drmessano: does it? |
| 03:45.26 | drmessano | Yep |
| 03:45.31 | [TK]D-Fender | drmessano: can it handel 4 calls with it? |
| 03:45.39 | drmessano | 4 calls? |
| 03:45.43 | drmessano | Its a 2 port ATA? |
| 03:45.48 | [TK]D-Fender | drmessano: 2 ports, 3-way each |
| 03:45.57 | drmessano | I dunno.. |
| 03:46.37 | drmessano | I just told you she puts out, now you wanna know what positions she likes.. Geez |
| 03:46.41 | drmessano | Call her yourself! |
| 03:46.42 | coppice | G.722.1 sounds better than G.722 most of the time, and that's at 32kbps |
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| 03:48.21 | drmessano | Interesting |
| 03:49.50 | drmessano | http://www.bswusa.com/proditem.asp?item=XSTREAM <-- 4 Grand just for 2 channels of G722 |
| 03:49.54 | drmessano | and a mixer |
| 03:50.00 | drmessano | over BRI.. bleck |
| 03:50.23 | drmessano | Although it will do IP, I can't remember the hoops needed to jump through |
| 03:51.18 | coppice | it does MP3 and AAC as well, but $4k sounds a "what the market will pay" price |
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| 03:51.36 | drmessano | coppice: No one uses the MP3 and AAC.. and yeah, thats about right |
| 03:51.47 | drmessano | 95% of broadcasters use G722 |
| 03:52.04 | drmessano | It saves time.. half the engineers can barely use the boxes to begin with |
| 03:52.26 | coppice | well, if they use it over ISDN, that would make sense. nothing else is properly standardised for use over ISDN |
| 03:54.03 | drmessano | But in case, that's the "standard" to replace.. For anything above POTS, the Zephyr w/BRI is the "high quality" path for remote broadcast, especially between two markets that have never heard of each other that suddenly need to send a show back/forth |
| 03:54.07 | drmessano | any* |
| 03:54.11 | coppice | [Tk]D-Fender have you ever compared a 64kbps MP3 with ulaw at the same rate? |
| 03:54.34 | drmessano | Anything less than $4000 that's easy to implement would rock |
| 03:55.02 | drmessano | I dont even work in broadcast anymore, but I still have cold sweats over those things |
| 03:55.14 | coppice | A cheap PC, a HFC BRI card, and a little software would do a great job |
| 03:55.33 | drmessano | Too complicated |
| 03:55.35 | [TK]D-Fender | coppice: So its what it is? Straight slight loss of a high enough rate to fit in the same and expand out to "somewhat better"? |
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| 03:55.41 | drmessano | You said "PC" |
| 03:55.55 | drmessano | Disc Jockeys need to be able to operate it |
| 03:56.25 | drmessano | I'm gonna start calling Linksys daily to find out when the G722 ATA is coming out |
| 03:57.04 | coppice | [Tk]D-Fender sorry, I can't decode that. MP3 at 64k does stereo at rather better than AM radio quality. ulaw sounds what it is - 100 year old telephone quality |
| 03:58.27 | coppice | but MP3 sucks for anything interactive, because of its algorithmic latency. that's why we have things like G.722.1 and CELT |
| 03:58.27 | drmessano | Improvements in codec quality are not so much magically making a bit stretch further, but moreso removing inefficiencies of past codecs |
| 04:00.05 | drmessano | Eventually you'll hit a point of dimishing returns, and by then, you won't have to worry about such low bandwidth paths.. |
| 04:00.18 | drmessano | But I think you'll still see improvement |
| 04:00.44 | coppice | there's no magic. for a general purpose codec you can strip out what the ear can't detect. for a specialist codec, like speech, you can strip out the ability to code what the source (voice) can't produce. we're still learning how to do that better, but we are reaching the limits. MP3 got impressively close for general purpose coding very early |
| 04:00.44 | drmessano | A lot of the codecs we use today were banged out so long ago, the algorithms are absolute crap |
| 04:00.46 | [TK]D-Fender | drmessano: And Hell will have frozen over.... but at least there'll be free skating ;) |
| 04:01.22 | [TK]D-Fender | drmessano: Is G.722 or derivatives patent encumbered? |
| 04:01.53 | drmessano | "we're still learning how to do that better" <--- That being the most significant part |
| 04:02.00 | coppice | yeah, but we're getting close to the limit in most areas |
| 04:02.09 | drmessano | Well |
| 04:03.06 | [TK]D-Fender | And IIRC, isn't G.723.1 supposed to have passed its patent epiry date? |
| 04:03.14 | coppice | you'll notice that for general purpose high latency coding, the last few years has produced only minor tweaks over MP3 |
| 04:03.29 | coppice | G.723.1 still has a while to go. 2014, I think |
| 04:03.35 | drmessano | By the time this generation of codecs is implemented in devices across the board, you'll have less people needing HD quality on low crap bandwidth links and attrition will bring more bandwidth and be more forgiving to higher bandwidth, higher quality codecs |
| 04:03.49 | drmessano | I dunno what patent issues G722 has, if any |
| 04:04.01 | coppice | G.722 is >20 years old |
| 04:05.02 | drmessano | Yeah |
| 04:05.51 | [TK]D-Fender | coppice: Wow... surprised I haven't seen it everywhere then.. |
| 04:06.05 | [TK]D-Fender | coppice: Why has it been so buried? |
| 04:06.28 | coppice | because it was design to replace G.711 on the PSTN, but BRI failed to take off |
| 04:06.32 | [TK]D-Fender | coppice: I only first heard of it 2-3 years ago |
| 04:07.09 | coppice | if you look in the bearer info for ISDN one option is "7.1kHz audio". That is for G.722 |
| 04:07.12 | drmessano | grumbles about it not failing completely enough |
| 04:07.46 | [TK]D-Fender | coppice: thats a lower sample rate than G.711 though... |
| 04:08.05 | coppice | 7.1kHz bandwidth. 16kHz sampling |
| 04:08.06 | drmessano | I haven't heard someone ask me about SPID's or DSN's in about a year. Good times. I could have another beer on that. |
| 04:08.36 | [TK]D-Fender | coppice: fewer samples, wider tone range = greater perceived quality? |
| 04:09.03 | coppice | huh? G.711 is 3.4kHz bandwidth. 8kHz sampling |
| 04:10.46 | [TK]D-Fender | coppice: Ok, I'll stop now before I lose any of us further. I am over my head on this and should simply go read up it. Anyone got a decent link or two? |
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| 04:25.40 | [TK]D-Fender | G.722 patents have expired, so it is freely available. |
| 04:26.12 | coppice | as I said, its >20 years old |
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| 04:28.52 | [TK]D-Fender | coppice: Thanks for the quick heads up, I'm going to continue mashing through Wikipedia to start and see how much sink in, then try to stay quiet for a while :) |
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| 04:49.37 | [TK]D-Fender | Alright checkout time, here's hoping I wake up NOT in convulsions tomorrow morning... |
| 04:49.39 | [TK]D-Fender | nite all |
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| 07:04.45 | syrusfrost | hi! |
| 07:06.58 | syrusfrost | I do a podcast, we take calls. Lots of wires, good stuff like that. I've googled a bit and cant find something that exactly matches my needs, but is there a scripting solution that would allow me to have call screeners take calls from an incoming queue and have them just hit a button to send them to the appropriate queue for answering on a live show? Keep in mind these call screeners would be working remotely... |
| 07:17.35 | drmessano | With the right handset, sure |
| 07:18.05 | drmessano | You have them answer a queue, then program a softkey to forward them to another queue |
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| 07:35.49 | syrusfrost | simple enough |
| 07:36.15 | syrusfrost | could you recommend a softphone that would handle that well? |
| 07:37.00 | syrusfrost | or rather, that is robust and easy to program |
| 07:38.42 | drmessano | Dunno about a softphone on that |
| 07:38.58 | drmessano | Would be much better to use a real phone |
| 07:42.21 | mchou | drmessano: you use tollfreegateway, right? |
| 07:42.54 | mchou | drmessano: do you mind calling an 800 # thru there and see if it's working for you? |
| 07:45.36 | drmessano | Hang on |
| 07:45.48 | drmessano | Newp |
| 07:46.12 | mchou | Newp? |
| 07:46.33 | mchou | as in no worky? |
| 07:46.43 | drmessano | Its not working, NO, NEWP, NADA, ZILCH, ZERO, NONYA, BORKY BORKEY BORKED |
| 07:46.48 | drmessano | Hmm |
| 07:47.10 | mchou | cool. it's not just me then :) |
| 07:47.12 | drmessano | Im gonna bounce one through IPKALL and see what happens |
| 07:48.21 | drmessano | IPKALL is teh LOL |
| 07:48.29 | drmessano | I sent it a tollfree call |
| 07:48.32 | mchou | what happened? |
| 07:48.46 | drmessano | It tells me it doesnt allow outbound calls except for tollfree numbers |
| 07:48.53 | drmessano | I CAN HAZ? |
| 07:48.54 | mchou | haha |
| 07:49.27 | mchou | I know tollfreegw is free but it's not the most reliable.... |
| 07:49.57 | mchou | maybe they are running asterisk on zune :) |
| 07:50.36 | drmessano | lol |
| 07:50.46 | drmessano | Well, theres several that are available with ENUM |
| 07:50.52 | drmessano | But one of them doesnt work at all |
| 07:50.57 | drmessano | and now this one is down |
| 07:51.27 | mchou | for now I'm using tf.callwithus.com |
| 07:51.37 | mchou | works |
| 07:52.30 | drmessano | I know a few of the providers offer free tollfree.. Les.net doesnt charge for my tollfree |
| 07:53.28 | mchou | yeah, but you probably need to register with then to dial toll free #s, right? |
| 07:53.42 | mchou | s/then/them |
| 07:54.28 | drmessano | Im not sure.. I would imagine so.. But I have noticed most providers terminate tollfree for free, even ones like Gizmo5 |
| 07:54.42 | drmessano | Still requires a signup though |
| 07:55.14 | mchou | drmessano: also, how exactly do you call a tollfree # witth ipkall? |
| 07:55.24 | drmessano | 18004664411@voiper.ipkall.com <-- APPARENTLY NOT A TOLLFREE CALL |
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| 07:55.50 | mchou | how about w/o 1? |
| 07:56.02 | drmessano | O.o |
| 07:56.10 | drmessano | I was wonderin'.. |
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| 07:56.56 | drmessano | Youre a fucking genius |
| 07:57.04 | mchou | lol |
| 07:57.08 | drmessano | Yep.. They trap the 1 |
| 07:57.25 | drmessano | I had thought about it for a second like 5 mins ago and dismissed it |
| 07:57.29 | drmessano | But hey |
| 07:58.36 | drmessano | Now I just need to make sure IPKALL doesnt go out of business |
| 07:58.36 | mchou | who knows |
| 07:58.59 | drmessano | Time to set up my "For a good time, call: " number on Craiglist |
| 07:59.03 | mchou | I think FCC or the tier 1 telcos put the squeeze on "settlements" |
| 07:59.25 | mchou | maybe ipkall is not long for this world |
| 07:59.50 | drmessano | Google needs to unzip their fly and do something here soon |
| 08:00.23 | mchou | drmessano: no joke. I was trying to dial tellme and dialed 800555TALK by accident |
| 08:00.42 | mchou | that turned out to be a sex chat line |
| 08:01.04 | drmessano | Googles damn "SIP coming soon" from 2004 is getting a bit stale |
| 08:01.13 | drmessano | Grandcentral is rotting on the vine |
| 08:01.19 | mchou | what? |
| 08:01.26 | mchou | GC works great for me |
| 08:01.35 | drmessano | Go sign up for another number |
| 08:01.38 | drmessano | Ill wait |
| 08:01.47 | mchou | I have a vested interest in GC not going out of Beta :) |
| 08:02.25 | mchou | cause once they go production they'll probably start charging |
| 08:02.25 | drmessano | Google bought it while it was in Beta.. It probably wont leave Beta or end up free |
| 08:02.26 | drmessano | Naw |
| 08:02.29 | drmessano | They may charge for a tier |
| 08:02.37 | drmessano | Like they do Google Apps |
| 08:02.50 | drmessano | Personal use free, business use xxx |
| 08:03.02 | mchou | maybe |
| 08:03.19 | drmessano | Google will put ads on it somewhere |
| 08:03.25 | mchou | but how mant #s you need from GC? |
| 08:03.31 | mchou | many* |
| 08:03.37 | mchou | 1 should be enough :) |
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| 08:04.12 | drmessano | I dont.. But its hardly anything to get friends or family using reliably when you have to wait 7 months for a week worth of signups |
| 08:04.48 | mchou | heh |
| 08:04.53 | drmessano | and right now, they've done ZERO with it in months.. which either means its being stripped down and integrated into GTalk, or they're just busy buying someone else |
| 08:05.41 | mchou | I dont know anyone who uses gtalk voice |
| 08:06.16 | drmessano | I would be happy if they offered DIDs for Gtalk accounts like a few of these places I surfed last night offer |
| 08:06.30 | drmessano | I would slap that on the * box and be done with it |
| 08:06.33 | syrusfrost | Gtalk+GC... wow |
| 08:08.52 | mchou | yeah, gtalk+GC would kill skype |
| 08:09.06 | syrusfrost | no doubt |
| 08:09.21 | drmessano | GC+Gizmo hasnt |
| 08:09.22 | syrusfrost | you said you had some gc #'s? |
| 08:09.22 | drmessano | heh |
| 08:09.43 | mchou | syrusfrost: who said that? |
| 08:09.49 | syrusfrost | lol nobody? |
| 08:10.12 | mchou | drmessano: gizmo is a bit brain dead |
| 08:10.34 | drmessano | mchou: I saw rigor mortis, optimist |
| 08:10.45 | mchou | lol |
| 08:11.15 | drmessano | Its great for being a replacment for whatever use USEDTOBE-FreeWorldDialup had |
| 08:11.15 | mchou | that's cause you didnt beat it |
| 08:11.19 | drmessano | but beyond, nah |
| 08:12.26 | syrusfrost | ~[ mchou ]~ alright, so you might not have spare GC number, but any info on GC? I'm too lazy to plow through thousands of requests in the nntp for a little info... |
| 08:12.53 | mchou | syrusfrost: nope. no inside scoop |
| 08:13.07 | mchou | syrusfrost: your guess is as good as mine |
| 08:13.16 | syrusfrost | poopy. I missed out man. |
| 08:14.04 | mchou | syrusfrost: you arent the only one |
| 08:15.22 | syrusfrost | oh I know, I envy you |
| 08:15.33 | mchou | lol |
| 08:15.57 | mchou | I got lucky is all |
| 08:16.11 | mchou | I was trying out voip at the time |
| 08:16.26 | mchou | had a buddy send me an invite |
| 08:16.33 | mchou | the rest is history |
| 08:17.10 | mchou | shortly after that GC closed the floodgates |
| 08:17.30 | drmessano | AH HA |
| 08:17.44 | drmessano | Predictions for Google's 2009 |
| 08:17.44 | drmessano | 4. GrandCentral will be publicly available in the US and the interface will integrate with Gmail. |
| 08:17.53 | drmessano | Off the Googlesystem blog |
| 08:18.00 | syrusfrost | oh wow, I'm looking at the reserve request screen and at this point all of the atlanta area codes are GONE |
| 08:18.00 | drmessano | Teaser or BS, dunno.. but a mention |
| 08:18.23 | syrusfrost | they were there before |
| 08:18.48 | drmessano | 11. Google Contacts will become a separate application, it will offer advanced search and an option to synchronize contacts data. |
| 08:19.30 | syrusfrost | It seems like they have an inventory going, I think they are serious about opening it up again... |
| 08:19.34 | syrusfrost | the question is when |
| 08:20.10 | drmessano | They are keeping up with the thousands of invites still floating around.. that is all |
| 08:20.59 | syrusfrost | I dont know man, all of the atlanta area codes gone? Those couldnt have all been invites could they? |
| 08:21.10 | syrusfrost | 678/404/770... |
| 08:21.33 | mchou | syrusfrost: where do you find out all area codes gone? |
| 08:21.55 | syrusfrost | http://www.grandcentral.com/home/reserve |
| 08:22.08 | syrusfrost | they were there before, I remember picking when I reserved |
| 08:22.43 | mchou | on snap |
| 08:22.51 | mchou | my area code is not there |
| 08:23.03 | mchou | how the hell is that possible |
| 08:23.19 | mchou | I'm like 2 ft. away from google HQ |
| 08:24.06 | syrusfrost | Google HQ? Even if all the employees took a number it's not possible that they were all eaten up |
| 08:24.19 | syrusfrost | they are collecting reservations and taking them out of inventory |
| 08:24.20 | mchou | that's my point |
| 08:24.33 | syrusfrost | that indicates intent to deliver imho |
| 08:24.57 | syrusfrost | it's not just a blind signup, some thought was put into this |
| 08:25.08 | mchou | maybe |
| 08:25.26 | syrusfrost | good point |
| 08:25.29 | mchou | but I know it's a level 3 exchange where I'm at |
| 08:25.45 | syrusfrost | nice |
| 08:26.25 | mchou | I wonder how many exchanges GC booked |
| 08:26.36 | mchou | cant be that many |
| 08:27.38 | syrusfrost | I'd love to see the numbers |
| 08:28.00 | mchou | yeah. same here |
| 08:29.55 | syrusfrost | ah man... speaking of level3 my employer just ditched their legacy broadwing network for at&t |
| 08:29.58 | syrusfrost | selloust |
| 08:30.03 | syrusfrost | sellouts* |
| 08:31.24 | syrusfrost | already seeing some lower quality with our cisco voice solutions |
| 08:34.09 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 08:42.15 | khronos | Hi guys. |
| 08:42.42 | khronos | What do you guys like for least cost dialing applications? |
| 08:42.57 | *** join/#asterisk dieguito84 (n=diego@87.18.187.20) |
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| 08:59.04 | *** join/#asterisk john_fbac (n=john_fba@216.186.221.211) |
| 09:07.10 | john_fbac | . |
| 09:09.41 | *** join/#asterisk john_fbac (n=john_fba@216.186.221.211) |
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| 09:33.08 | *** join/#asterisk qdk (n=qdk@94.191.236.28.bredband.3.dk) |
| 09:40.39 | EmleyMoor | For a few days I hae been unable to enable/disable polling on my DVD writer drive - it is otherwise functional though. hal-disable-polling soays it cannot find the device. Why would that be? |
| 09:41.37 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
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| 09:56.43 | Madkiss | hi all. i am trying to set up hylafax=>iaxmodem via chan_lcr. the iax-stuff seems to be set up correctly, but actually, hylafax reports a "no carrier"-error |
| 10:04.39 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:05.44 | Madkiss | okay, got it. strike. |
| 10:08.11 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
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| 10:37.49 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-79-207.w86-215.abo.wanadoo.fr) |
| 10:41.06 | verywiseman | i have linksys 3102 , how can i define its fxo fxs port to asterisk? |
| 11:05.16 | *** join/#asterisk sosoriri (n=chatzill@222.47.180.130) |
| 11:14.20 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) [NETSPLIT VICTIM] |
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| 11:14.21 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 11:15.08 | verywiseman | i have linksys 3102 , how can i define its fxo fxs port to asterisk? |
| 11:28.45 | *** join/#asterisk reneger (n=reneger@p3EE2E5C3.dip.t-dialin.net) |
| 11:33.34 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 11:37.42 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 11:53.28 | delcoyote | hi |
| 11:54.58 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 11:59.43 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 12:10.09 | *** join/#asterisk adminguru (n=atze@p57BD608A.dip.t-dialin.net) |
| 12:43.33 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 13:13.46 | Madkiss | is an overview over the various asterisk variables available? e.g. i would like to know whether there is some variable that tells me who accepted a call. i.e. it rings, somebody picks the call up and i would like to store that information somewhere. |
| 13:14.18 | mmlj4 | the old handbook listed all of those |
| 13:14.41 | Madkiss | oh I see |
| 13:18.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 13:33.47 | PanGoat | Does the bot work for everyone? or is does it have a limited set of nicks it will allow commands from? |
| 13:33.53 | PanGoat | ~book |
| 13:34.16 | jbot | methinks book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 13:34.29 | PanGoat | coooool |
| 13:42.27 | *** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
| 13:46.42 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 14:03.15 | *** join/#asterisk oh207 (n=oh207@nylug/member/oh207) |
| 14:08.34 | *** join/#asterisk neurosys (n=vinix@c-71-196-16-43.hsd1.fl.comcast.net) |
| 14:14.44 | *** join/#asterisk micols (n=micols@scharff.fys.ku.dk) |
| 14:14.55 | micols | WARNING[20695]: app_dial.c:1111 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) anyone know this error and possible fix for it? |
| 14:15.09 | micols | restart asterisk? |
| 14:17.43 | micols | i keep getting it for all outgoing calls |
| 14:18.36 | kannan | micols - chan_zap.so is loaded ok? |
| 14:18.50 | micols | zap show channels shows up fine |
| 14:20.04 | unixdawg | but does it show channels |
| 14:20.45 | unixdawg | are your ports showing up |
| 14:22.29 | micols | http://nopaste.com/p/a7bS78p0r |
| 14:23.20 | micols | i notice a few RED alarms, so i guess the congestion is due to that? |
| 14:30.41 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 14:31.00 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 14:31.48 | micols | yes it has RED alarms on 3 of 4 cards, so i will try to restart the PBX, it worked last time i think. |
| 14:47.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:56.07 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 14:56.24 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 15:28.09 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
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| 15:37.46 | *** part/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 15:53.12 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 15:53.17 | eppigy | hello |
| 15:53.19 | eppigy | i am dave |
| 15:54.41 | postel | OMG Hi!!!111 |
| 15:55.46 | eppigy | 8[] |
| 16:00.28 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 16:06.43 | drmessano | Om nom nom? |
| 16:08.20 | eppigy | possibly |
| 16:09.53 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.29) |
| 16:11.23 | Daejeo | can anyone recommend a machine for handling 100 calls simultaneously ? |
| 16:12.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:13.48 | drmessano | Dual Quad Core 3GHZ Xeons and 8GB Ram |
| 16:14.03 | eppigy | dang |
| 16:14.39 | eppigy | Daejeo: compression, no compression, dahdi channels? |
| 16:14.45 | eppigy | i mean lets get real here |
| 16:16.49 | Daejeo | eppigy: g729 |
| 16:17.06 | drmessano | goes back and repunts |
| 16:17.08 | drmessano | Dual Quad Core 3GHZ Xeons and 8GB Ram |
| 16:17.31 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-79-207.w86-215.abo.wanadoo.fr) |
| 16:17.31 | Daejeo | drmessano: thank you |
| 16:18.07 | eppigy | BOOYA |
| 16:18.39 | Daejeo | does DELL sell? |
| 16:18.46 | drmessano | They do sell |
| 16:18.50 | drmessano | 24/7 I hear |
| 16:18.50 | eppigy | WELL THEY ARENT GIVING THEM AWAY |
| 16:18.53 | drmessano | GIVE IT TO THEM |
| 16:19.34 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:20.17 | Daejeo | any model number? |
| 16:21.12 | eppigy | poweredge 1950 gen3 |
| 16:21.16 | jaytee | ewwww, these Commit Cappucino nicotine lozenges taste like ass |
| 16:22.32 | drmessano | Trying to quit? |
| 16:22.56 | jaytee | yeah, thank god this was a fee sample |
| 16:22.59 | jaytee | free |
| 16:23.01 | eppigy | bro |
| 16:23.04 | drmessano | FEE SAMPLE |
| 16:23.05 | eppigy | here is some advice |
| 16:23.11 | eppigy | get some advantix |
| 16:23.13 | drmessano | LIKE FEEWORLDDIALUP? |
| 16:23.16 | eppigy | lol |
| 16:23.23 | eppigy | I quit 2 years ago |
| 16:23.28 | jaytee | advantix? is that like Chantix? |
| 16:23.29 | drmessano | From Jeff Pulverized.com |
| 16:23.34 | eppigy | chantix |
| 16:23.35 | eppigy | lol |
| 16:23.36 | eppigy | yes |
| 16:23.39 | eppigy | advantix |
| 16:23.45 | eppigy | where the hell did i get that from |
| 16:23.48 | drmessano | I used the Feline Advantix |
| 16:23.59 | jaytee | I'm not taking any SSRI based meds ever again |
| 16:24.02 | drmessano | Now I am Flea Free |
| 16:24.14 | eppigy | but nicotine supplements just prolong your withdrawal |
| 16:24.20 | eppigy | and you will never quit |
| 16:24.43 | drmessano | jaytee: Take some Wellbutrin, a shot of jim beam/coke, and pop a Mucinex |
| 16:24.49 | eppigy | lol |
| 16:24.49 | drmessano | Thats a weeks vacation |
| 16:24.58 | eppigy | just get soem advantix |
| 16:25.02 | jaytee | probably true, but if I have a craving and use gum or a lozenge at least I'm not giving myself the carbon monoxide and tar that goes with inhaling smoke. |
| 16:25.02 | eppigy | i mean chantix |
| 16:25.09 | eppigy | it works like a charm |
| 16:25.14 | eppigy | and its relatively cheap |
| 16:25.21 | drmessano | God damnit |
| 16:25.26 | drmessano | I have some stupid cats |
| 16:25.31 | eppigy | choke them |
| 16:25.33 | eppigy | to death |
| 16:25.34 | jaytee | you too! I have two of them |
| 16:25.40 | drmessano | I now have THREE pencils with NO erasers |
| 16:25.42 | jaytee | fur covered retards, both |
| 16:25.44 | eppigy | D: |
| 16:25.46 | *** join/#asterisk voxter (n=voxter@190.10.4.153) |
| 16:25.51 | eppigy | i hate cats so much |
| 16:25.53 | drmessano | Sticking out of a coffee mug/pen holder |
| 16:25.55 | eppigy | i9m not gonna lie |
| 16:26.00 | drmessano | and they chew them off |
| 16:26.06 | drmessano | CARBON SIDE UP, BITCHES |
| 16:26.20 | drmessano | CHEW THAT |
| 16:26.24 | jaytee | lol |
| 16:26.36 | drmessano | They are dumb |
| 16:26.41 | drmessano | I put a WRT54G on the floor |
| 16:26.49 | *** join/#asterisk propellerhead (n=yogurt2u@host11.190-136-245.telecom.net.ar) |
| 16:26.50 | drmessano | 12 hours later, antennas chewed up |
| 16:26.52 | drmessano | LIKE DOGS DO |
| 16:26.55 | drmessano | NOT CATS |
| 16:26.56 | jaytee | other than using CDR which won't give me realtime stats is there anyway to monitor active channels for concurrent calls? |
| 16:26.56 | eppigy | and they installed linux on it? |
| 16:27.05 | drmessano | eppigy: Damn right |
| 16:27.15 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:27.17 | drmessano | ~book |
| 16:27.18 | jbot | from memory, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 16:27.28 | drmessano | ^^^^^^^^^^^^^^^^^^ READ AND STFU4EVA |
| 16:27.33 | drmessano | Sorry |
| 16:27.37 | eppigy | jaytee: there is a cool zenoss plugin |
| 16:28.15 | eppigy | also nagios of course |
| 16:28.20 | eppigy | if you just want alerts |
| 16:28.27 | eppigy | when you need to scale up |
| 16:29.05 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 16:33.31 | jaytee | eppigy, I just want to be able to see how many calls are in progress at any given time. sip show inuse gives me nothing, even when I know a call is in progress |
| 16:34.34 | eppigy | this may e a dumb question but what about sip show channels? |
| 16:35.00 | jaytee | duh!!! god, I'm an idiot!!! |
| 16:35.50 | eppigy | 8[] |
| 16:37.22 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
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| 16:47.38 | *** join/#asterisk [netman] (n=netman@188.Red-88-23-81.staticIP.rima-tde.net) |
| 16:51.19 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 16:51.58 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 16:53.39 | mmlj4 | sip show channels |
| 16:53.49 | *** join/#asterisk suvir (n=suvir@ppp-124-120-129-69.revip2.asianet.co.th) |
| 16:54.52 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:05.38 | *** join/#asterisk DelphiWorld (n=Miranda@41.221.26.251) |
| 17:05.53 | DelphiWorld | hi my friend |
| 17:06.17 | DelphiWorld | what is the best free linux PBX Distro ? |
| 17:07.19 | jaytee | do you mean what is the best linux distro to run a PBX application like Asterisk on? opinions are wide and varied. |
| 17:07.20 | drmessano | Linux |
| 17:08.00 | DelphiWorld | jaytee: mor linux distro is a specific for asterisk |
| 17:08.08 | DelphiWorld | each one ? |
| 17:08.16 | jaytee | none are specific to Asterisk |
| 17:08.56 | UQlev | DelphiWorld: best linux distro is Gentoo |
| 17:08.59 | DelphiWorld | then what is trixbox ? |
| 17:09.17 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:09.57 | jaytee | trixbox is asterisk at the core with a forked version of the freepbx gui and a few other addons running on top of linux. |
| 17:10.45 | DelphiWorld | jaytee: then trixbox is a full linux dsitro specific to asterisk ? |
| 17:11.36 | drmessano | Trixbox is not a linux distro |
| 17:11.40 | jaytee | no, trixbox is a VoIP pbx solution using asterisk code. It will run on lots of different linux distros. They just bundle the whole thing on one particular distro |
| 17:12.01 | drmessano | Its CENTOS with some fancy ks config settings to install Trixbox RPMs out of the box |
| 17:12.24 | DelphiWorld | jaytee: i understand now |
| 17:12.32 | drmessano | At one point it was a giant tarball |
| 17:12.49 | DelphiWorld | and any asterisk win32 implementation (please except for "asterisk win32") ? |
| 17:13.01 | drmessano | and it got too complex for them to manage on different systems, so now its distributed as a CentOS CD with the tarball and configs |
| 17:13.02 | jaytee | DelphiWorld, if you want more info about trixbox go ask in the #trixbox channel. If you're just starting to learn this stuff I recommend avoiding trixbox and using asterisk without a GUI and run it on CentOS or Debian. |
| 17:14.20 | DelphiWorld | jaytee: please any asterisk port to windows ? |
| 17:14.27 | drmessano | lol |
| 17:14.39 | jaytee | DelphiWorld, you don't even want to go there!!! |
| 17:14.59 | drmessano | Symantec Antivirus for Asterisk PBX 1.0 <--- |
| 17:15.06 | drmessano | No |
| 17:15.06 | jaytee | lol |
| 17:15.10 | drmessano | Thats too 2002 |
| 17:15.37 | drmessano | Symantec Endpoint Protection for Asterisk PBX Systems 1.0 |
| 17:16.00 | DelphiWorld | jaytee: please note that i'm a blind user |
| 17:16.07 | drmessano | "The SIP security you need to slow down the flow of malware, call quality, and business growth" |
| 17:16.17 | jaytee | DelphiWorld, I'll take that into consideration |
| 17:16.59 | DelphiWorld | i'm asking about graphical USER Interface bicose i'm using windows only (with a screen reader) |
| 17:18.27 | riddlebox | theres always putty from a windows box |
| 17:18.47 | *** join/#asterisk soa2ii (n=soa2ii@i59F579E6.versanet.de) |
| 17:18.54 | drmessano | cues up Billy Mays |
| 17:18.58 | drmessano | ITS WONDER PUTTY |
| 17:19.09 | riddlebox | lol |
| 17:19.39 | drmessano | I am going to HANG this entire PBX from the wall with ONE PEA SIZED drop of WONDER PUTTY |
| 17:20.05 | riddlebox | i just saw a commercial for his wax |
| 17:20.06 | *** join/#asterisk ManxPower (n=manxpowe@176.sub-75-254-174.myvzw.com) |
| 17:20.13 | drmessano | See this bond? Sure, its stuck to a thin sheet of paper on the outside of the sheetrock, but it will hold for YEARS |
| 17:20.28 | soa2ii | If I want to connect my home pc to the telephone net what hardware do I need? Smth like a "FRITZ!Card PCI" should fit, doesn't it? |
| 17:20.51 | drmessano | Gulp |
| 17:20.52 | ManxPower | soa2ii: Does your home PC run Linux? |
| 17:21.01 | soa2ii | ManxPower: Yeah. |
| 17:21.26 | ManxPower | soa2ii: You can use a PCI card or you can subscribe to an ITSP. What kind of phone line do you currently have? |
| 17:22.18 | soa2ii | ManxPower: ISDN with telephone flat. So I don't want to subscribe smth else... I just want to "tunnel" VoIP to the "real" telephone net :P |
| 17:22.25 | ManxPower | If you just want cheap phone calls I recommend a Linksys/SIPura box and an ITSP. |
| 17:22.36 | Daejeo | can anyone recommend the best E1 pci card ? |
| 17:22.45 | ManxPower | Daejeo: Digium or Sangoma |
| 17:22.46 | *** join/#asterisk zr0 (i=br@unaffiliated/zr0) |
| 17:23.03 | zr0 | does asterisk support T.38? |
| 17:23.20 | soa2ii | ManxPower: Well... it should later work over VPN from everywhere in the world without carriing hardware and so... |
| 17:23.23 | ManxPower | soa2ii: you are not going to connect to the "telephone net" (we call that the PSTN) unless you subscribe to some service. |
| 17:23.31 | Daejeo | ManxPower: both are same? |
| 17:23.39 | soa2ii | So I would like to connect my home server to the isdn box. |
| 17:23.48 | ManxPower | Daejeo: No. One is made by Digium and one is made by Sangoma? |
| 17:23.48 | drmessano | ... |
| 17:24.00 | Daejeo | I meant quality |
| 17:24.02 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:24.21 | soa2ii | ManxPower: So it is not possible to tell asterisk it should call some "normal" phone over my isdn? |
| 17:24.42 | ManxPower | soa2ii: Sure it is, but you have to subscribe to the ISDN. |
| 17:24.56 | zr0 | isdn is expensive |
| 17:25.03 | ManxPower | I don't know much about ISDN BRI and Asterisk since it's not supported in the USA (where I live) |
| 17:25.20 | ManxPower | zr0: ISDN is priced location to location. It is expensive some places it is cheaper than analog in other places. |
| 17:25.31 | soa2ii | ManxPower: Well I have of course a running internet/isdn flatrate here... I just want to make my computers possible to call "real" phones :P |
| 17:25.36 | zr0 | ManxPower: you mean in europe its cheaper |
| 17:25.59 | ManxPower | zr0: in europe an IDSN BRI is cheaper than 2 analog lines in many places. |
| 17:26.18 | ManxPower | Also in Tennessee, but most of the USA it is much more expensive. |
| 17:26.25 | zr0 | right :) |
| 17:26.28 | ManxPower | soa2ii: usually flat rate is only for voice calls. |
| 17:26.47 | ManxPower | soa2ii: buy an ISDN BRI card, install Asterisk, be happy. |
| 17:26.47 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:27.01 | zr0 | isdn lines have great call quality |
| 17:27.05 | soa2ii | ManxPower: OK. What ISDN BRI do you recommend? |
| 17:27.19 | zr0 | one that works with asterisk? |
| 17:27.28 | soa2ii | zr0: Shure |
| 17:27.34 | ManxPower | soa2ii: I assume what ISDN BRI card do I recommend and I just said I don't know much about the cards because I can't use them |
| 17:27.54 | soa2ii | Aww... ok (: |
| 17:28.00 | zr0 | soa2ii: just go with whatever digium recommends |
| 17:28.11 | soa2ii | digium? |
| 17:28.15 | florz | soa2ii: any cologne chip (HFC-S PCI A) will do |
| 17:28.15 | ManxPower | soa2ii: there is no official ISDN BRI card that is well known and supported. Digium has one card that uses some form of mISDN (zaptel support coming) |
| 17:28.38 | ManxPower | the Digium card is pretty recent and not well tested by users in my opinion. |
| 17:28.49 | soa2ii | Hm. |
| 17:29.49 | ManxPower | ~mailinglist |
| 17:29.50 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 17:29.50 | soa2ii | What do I need furthermore? Do I need an own telephone number for my server then? |
| 17:29.53 | ManxPower | I would suggest you search the mailing lists, but jbot seems to be asleep |
| 17:30.06 | ManxPower | soa2ii: you are plugging your existing line into Asterisk. What else do you expect to need? |
| 17:30.08 | soa2ii | Thanks. |
| 17:30.25 | ManxPower | soa2ii: also read The Book |
| 17:30.27 | ManxPower | ~book |
| 17:30.27 | jbot | book is probably probably 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://tfot.leifmadsen.com or see ~buybook |
| 17:30.29 | soa2ii | ManxPower: Well... I say more about my setup. |
| 17:30.55 | ManxPower | PSTN -> ISDN BRI -> BRI PCI card -> Asterisk. Pretty simple. |
| 17:31.25 | soa2ii | I have the isdn splitter. then my internet router and one isdn phone with two numbers connected to the splitter. now I just want to add my server next to the existing phone (: |
| 17:31.42 | zr0 | ManxPower: do you have any experience faxing in asterisk over voip? |
| 17:32.32 | ManxPower | zr0: Yes. It doesn't work. That is my experience. |
| 17:32.50 | ManxPower | soa2ii: you are not going to be able to do that with ISDN. |
| 17:33.22 | soa2ii | ManxPower: Why? |
| 17:33.40 | ManxPower | it sounds like what you have is an ISDN line connected into a router and a device to provide analog phone ports. |
| 17:34.03 | ManxPower | soa2ii: because your internet will stop working when you unplug the ISDN line from your router and NT1 and plug it into Asterisk |
| 17:34.37 | soa2ii | ManxPower: I want to plug my computer in like an ISDN telephone |
| 17:34.47 | soa2ii | simply next to the existing phone |
| 17:34.53 | soa2ii | this is impossible? |
| 17:35.04 | zr0 | ManxPower: do you think it would work with a real rax machine connected through an ata? |
| 17:35.07 | ManxPower | soa2ii: I doubt that is possible, but you should look at the mailing lists. |
| 17:35.32 | ManxPower | zr0: the only reliable way to fax with Asterisk in my experience is to not fax thru asterisk and fax thru an analog line direct from your telco |
| 17:35.43 | soa2ii | Hmmmm |
| 17:36.02 | ManxPower | soa2ii: Asterisk is designed as a PBX. PBXs don't share phone lines with external devices. |
| 17:36.26 | zr0 | ManxPower: bummer |
| 17:38.14 | *** join/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:38.53 | ManxPower | zr0: Other people's experiences may be different |
| 17:39.37 | zr0 | ManxPower: when you tried it, where you using a real fax machine on your end? |
| 17:41.34 | yang | zr0: I had some good results in receiving faxes over SIP |
| 17:41.50 | ManxPower | zr0: I have used faxing in three ways in Asterisk. 1) analog line -> Asterisk -> analog fax. Didn't work well. 2) analog line -> analog fax Never had a problem. 3) PRI -> Asterisk -> app_rx_fax Works most of the time, but we have analog fax when there are problems |
| 17:43.19 | zr0 | hmm. i'm trying to do analog line -> voip gateway -> my pbx -> ata -> my fax machine, which sounds more complicated then what you were doing |
| 17:43.50 | zr0 | i understand that the t.38 protocol can alleviate a lot of problems |
| 17:44.02 | zr0 | yang: did you have a t.38 provider? |
| 17:44.15 | soa2ii | ManxPower: So again... that you're shue what I want to do: http://www.lagom.de/misc/setup.png |
| 17:44.23 | yang | zr0: no |
| 17:45.03 | yang | zr0: I have difficulties in sending from fax machine over ATA, but received faxes go in proper to asterisk/hylafax |
| 17:45.30 | zr0 | yang: so this is straight over a sip trunk using ulaw? |
| 17:45.51 | [TK]D-Fender | zr0: What is this "voip gateway" you're speaking of? |
| 17:46.04 | yang | zr0: yep |
| 17:46.26 | zr0 | [TK]D-Fender: whomever transfers the pstn call over to my sip trunk |
| 17:47.01 | [TK]D-Fender | zr0: never write those 2 words like that for your case then. put (ITSP). |
| 17:47.14 | [TK]D-Fender | zr0: Your way made it look like YOU had an analog line |
| 17:47.20 | zr0 | ok. what does that stand for? |
| 17:47.29 | [TK]D-Fender | zr0: and were converting it locally with your own equipment |
| 17:47.33 | [TK]D-Fender | ~itsp |
| 17:47.34 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:47.35 | [TK]D-Fender | ^^^^^^^^^ |
| 17:47.51 | zr0 | thanks |
| 17:47.55 | coppice | most ATAs have settings for FAXing with Alaw/ulaw, but they are really bogus. their chance of working is small, even when packet loss is very low. faxing into an iaxmodem/hylafax setup generally gives good results, if the packet loss os very low |
| 17:48.46 | zr0 | coppice: that's the impression i've got, but there are atas out there that claim to implement t.38 over ulaw |
| 17:49.13 | [TK]D-Fender | coppice: I'd head that in RTP where 20ms packets are the norm even 1 lost packet risks the entire call. How accurate is that? |
| 17:49.24 | coppice | er, no. T.38 does not run over ulaw. you use T.38 instead of ulaw |
| 17:49.24 | [TK]D-Fender | heard* |
| 17:49.31 | zr0 | right, sorry |
| 17:49.57 | zr0 | coppice: do you know if asterisk supports t.38? |
| 17:50.14 | *** part/#asterisk unixdawg (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:50.46 | coppice | [TK]D-Fender: well, in theory things should retry, but in practice they can get very flaky even with very few lost packets. the big problem with ulaw FAXing with ATAs is jitter, though. they fall apart with even modest amounts of jitter |
| 17:51.33 | coppice | zr0: my T.38 stuff is being integrated into *, but I don't know the current status |
| 17:51.44 | [TK]D-Fender | coppice: Sonds like fun. Diet-Fun... jsut like Real-Fun, only half as much.... |
| 17:51.49 | zr0 | which makes sense, you're adding so much latency in |
| 17:52.27 | coppice | latency is not the big issue. jitter is |
| 17:52.58 | zr0 | i mean, latency at the ip level |
| 17:53.49 | coppice | latency is not the big issue. jitter is |
| 17:53.53 | [TK]D-Fender | zr0: Yes, * 1.4 supports it in passthrough which is what you'll need for your ATA. |
| 17:54.11 | zr0 | [TK]D-Fender: badass. thank you. i think i'm gonna try this out. |
| 17:54.18 | [TK]D-Fender | zr0: That is if you get one that supports it. Does your ITSP claim T.38 support? |
| 17:54.43 | zr0 | [TK]D-Fender: that's the next thing to find out. i'm pretty sure they don't. :) |
| 17:55.02 | zr0 | [TK]D-Fender: but it's not a big deal for me to switch providers |
| 17:55.33 | coppice | zr0: do you need a real FAX machine, or is computer FAXing appropriate for you? |
| 17:56.09 | [TK]D-Fender | zr0: And pay extreme heed to coppice's warnings & advice |
| 17:56.32 | zr0 | coppice: i want to use a real fax machine |
| 17:56.52 | zr0 | coppice: to send faxes |
| 17:57.05 | zr0 | coppice: more than to receive |
| 17:58.52 | zr0 | [TK]D-Fender: by pass-thru, does that mean i need to have a seperate did dedicated to t.38? |
| 18:02.03 | eppigy | hello |
| 18:04.31 | [TK]D-Fender | zr0: It means that if your ATA supports T.38 and your ATA does, then * will let them talk it though and do their thing. |
| 18:04.50 | [TK]D-Fender | zr0: * does not have to translate or decode T.38 itself. |
| 18:04.56 | [TK]D-Fender | eppigy: You are dave!!!!!!!!!!!!!!! |
| 18:06.47 | eppigy | YES |
| 18:06.55 | eppigy | I AM HE WHO IS KNOWN AS DAVE |
| 18:07.35 | jaytee | hehehee |
| 18:08.51 | eppigy | YEAH SON |
| 18:13.36 | eppigy | what |
| 18:16.13 | *** join/#asterisk SpecialEd (n=wut@cpe-72-179-194-139.stx.res.rr.com) |
| 18:26.35 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 18:34.49 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 18:37.45 | drmessano | Dave is such a stupid name |
| 18:37.55 | justdave | I resent that |
| 18:38.07 | drmessano | Mainly because the 2 Dave's I have known in any real capacity were douches |
| 18:38.07 | justdave | er, resemble that, or something |
| 18:38.26 | drmessano | One tried to get me kicked off a hockey team |
| 18:38.32 | drmessano | The other stole $50 from me |
| 18:38.38 | drmessano | Not had a good "dave" experience |
| 18:38.46 | drmessano | So screw "Dave's" |
| 18:38.54 | drmessano | So screw "Dave"'s |
| 18:38.56 | drmessano | There |
| 18:38.58 | drmessano | or somehting |
| 18:39.02 | eppigy | MAN |
| 18:39.04 | eppigy | man |
| 18:39.14 | eppigy | dont be dave jaded |
| 18:39.20 | eppigy | because of two bad apples |
| 18:39.41 | drmessano | I don't want to be Dave jaded.. But the statistics |
| 18:40.04 | drmessano | Do you know guys named "Dave" are 3 times more likely to be killed as innocent bystanders in bar fights? |
| 18:40.09 | drmessano | THE DECK IS STACKED |
| 18:40.23 | justdave | only because there's three times as many Daves as other names :) |
| 18:40.39 | eppigy | you have to calulate per capita |
| 18:40.45 | drmessano | My reference "God I miss Dave. If only he had not gone to that bar.." |
| 18:40.46 | eppigy | lets be real here |
| 18:41.30 | drmessano | Dave sounds like such an innocent name.. Like a Dave could do no wrong, but man, shame he was in the wrong place at the wrong time. |
| 18:41.51 | drmessano | "If only Dave had waited 30 more seconds before going to work" |
| 18:42.39 | drmessano | "Why was Dave at the mall on a Sunday!!!! He never goes... :(" |
| 18:42.39 | drmessano | READ THE NEWSPAPERS PEOPLE |
| 18:42.51 | eppigy | lies |
| 18:43.02 | drmessano | Not only are Dave's more likely to be douchebags, but they apparently have really bad luck |
| 18:43.10 | eppigy | LIES |
| 18:43.50 | drmessano | "Dave was such a nice guy. All he was doing was mowing his elderly neighbors lawn. You never would expect this to happen..." |
| 18:44.03 | drmessano | I almost feel bad for "Dave" |
| 18:44.10 | drmessano | But then I look at my wallet |
| 18:44.21 | drmessano | Yep.. $50 less in there.. 15 years running |
| 18:44.35 | drmessano | I WANT MY $50 BACK |
| 18:45.21 | jaytee | lol |
| 18:45.42 | eppigy | get it from his estate |
| 18:45.48 | Dovid | jaytee: where u here a fay days ago when i was speaking to the dev.'s ? |
| 18:46.43 | [TK]D-Fender | Dovid: Careful.... the Grammar Rangers will get you in your sleep.... |
| 18:47.36 | jaytee | Dovid, I was never here on fay days. Fay days are a religious holiday for me and there are two things I never ever do. One is come into chat on fay days and two, "I never roll on Shabbos" |
| 18:47.39 | eppigy | syntax commandos |
| 18:48.04 | Dovid | TK: been a long day...... |
| 18:48.25 | eppigy | full of fat rails? |
| 18:48.47 | Dovid | jaytee: Not here on Shabbat either ;) |
| 18:48.54 | jaytee | but to be serious (god I hate this) I don't recall when you were talking to the devs. What was it about? |
| 18:49.21 | Dovid | about having a context that would be called for every other context with out usign include |
| 18:49.51 | eppigy | like a broadcast context |
| 18:49.59 | eppigy | WHAT AN INCREDIBLE CONCEPT |
| 18:50.03 | jaytee | or a "universal context" |
| 18:50.11 | Dovid | sort of |
| 18:50.14 | [TK]D-Fender | Dovid: Stupidly dangerous, and a very wrong approach to the fact you can't code... |
| 18:50.17 | jaytee | or The God Context |
| 18:50.23 | drmessano | Wait Wait |
| 18:50.29 | Dovid | TK: Which is y i pad a dev. to do it |
| 18:50.30 | drmessano | [general] <-- |
| 18:50.37 | drmessano | ZOMG COMMIT |
| 18:50.42 | eppigy | Dovid: how would you calculate the broadcast context given the subnet? |
| 18:50.49 | Dovid | sorry. nto broadcast |
| 18:50.53 | [TK]D-Fender | has problems with commitment... |
| 18:50.54 | Dovid | not* |
| 18:50.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:50.57 | eppigy | drmessano: ^5 |
| 18:51.08 | jaytee | I'm gonna defer to [TK]D-Fender on this one, it opens way too many cans of worms and 1) I've never seen cans of worms sold anywhere, even in bait stores and 2) I don't feel like going fishing at the moment. |
| 18:51.38 | drmessano | What is wrong with "Include"..? |
| 18:51.39 | Dovid | jaytee: http://bugs.digium.com/view.php?id=14159 |
| 18:51.41 | drmessano | Lazyiness? |
| 18:51.54 | drmessano | -y |
| 18:52.02 | jaytee | wtf? that took me to a gay porn site!!! |
| 18:52.06 | [TK]D-Fender | Worms are tpically sold in lidless styrofoam containers full of earth |
| 18:52.18 | Dovid | drnessano: I was dealing with the gui (which I had no coice but needed to use). and it would over write what i put in there |
| 18:52.46 | jaytee | [TK]D-Fender, yes never in cans which begs the question where the expression came from in the first place. Obviously someone that never fished much. |
| 18:52.46 | drmessano | Dovid: If it was FreePBX, youre doing it wrong |
| 18:52.55 | drmessano | Dovid: Easy to work around |
| 18:53.15 | Dovid | the asterisk gui. not freepbx |
| 18:53.15 | [TK]D-Fender | Dovid: Trying to correct the fact a GUI is screwing you in the ass? STOP USING IT |
| 18:53.19 | Dovid | i HATE gui's |
| 18:53.37 | eppigy | you have never seen illustrations of a used soup tin full of worms? |
| 18:53.41 | Dovid | [TK]D-Fender: My dman boss !!! most of time times he listens to me |
| 18:53.42 | [TK]D-Fender | Dovid: You don't rewrite * to work around some retard GUI. |
| 18:53.42 | drmessano | Stop pounding me in the ass.. File a bug report with the GUI |
| 18:53.48 | drmessano | Dont try to fix Asterisk |
| 18:53.52 | drmessano | Its not broken |
| 18:54.00 | drmessano | and globalcontexts would be an ENABLED |
| 18:54.01 | drmessano | and globalcontexts would be an ENABLER |
| 18:54.13 | drmessano | Like having more options than NAT=YES |
| 18:54.17 | [TK]D-Fender | drmessano: think about.... CONTEXT CONTENTION! |
| 18:54.20 | Dovid | drmessano: The GUI is made to be simple (One of the asterik dev's told me that) |
| 18:54.23 | drmessano | NAT=NAT <--- Would cover it all |
| 18:54.26 | eppigy | NAT=MAYBE |
| 18:54.42 | jaytee | and the one time you find yourself needing a context that doesn't automatically inherit that context then whaddaya do? Punt? |
| 18:54.42 | [TK]D-Fender | Dovid: Prepare for supreme F-ups because of include order prioritization. |
| 18:54.47 | eppigy | NAT=GODIAMNOTGOODWITHNETWORKSHOWDIDIGETHERE |
| 18:55.05 | drmessano | NAT=yes doesnt mean yes, no doesnt mean no, never means no, and always means never, and no means maybe, and yes means sometimes and OMG WHAT THE FUCK WHO WROTE THIS? |
| 18:55.09 | [TK]D-Fender | Dovid: there are other ways to solve your problem. |
| 18:55.10 | Dovid | [TK]D-Fender: 100%. i just use it for the h extension |
| 18:55.23 | eppigy | TRIPPIN |
| 18:55.47 | [TK]D-Fender | Dovid: I'm sorry, since when was * all about YOU? |
| 18:56.01 | drmessano | ASTERISK IS ABOUT YOU |
| 18:56.02 | tzafrir_laptop | Dovid, generally 1.4 is not getting any new features. If you actually want that feature accepted, please provide a patch vs. trunk |
| 18:56.03 | [TK]D-Fender | dovoPut. Down. The. Crack. Pipe. (c) JerJer |
| 18:56.04 | drmessano | ITS YOUR PBX |
| 18:56.09 | Dovid | [TK]D-Fender: It isn't. which is why i paid some one to create it for me |
| 18:56.16 | drmessano | ASTERISK and YOU.... Probably |
| 18:56.21 | tzafrir_laptop | That said, I'm not sure how useful it is, considering the alternatives |
| 18:56.31 | drmessano | You... paid... someone... for.... that.... patch? |
| 18:56.46 | Dovid | tzafrir_laptop: you may be right but didnt hurt to give it to others. |
| 18:56.53 | drmessano | puts.... down..... the....... pointy.... must.... not..... gouge.... own....... eyes |
| 18:56.55 | tzafrir_laptop | Right |
| 18:57.08 | Dovid | drmessano: When I have been yp for days and need a quick fix I throw money at my problems |
| 18:57.26 | [TK]D-Fender | Dovid: Oh yes, someone might go code it... it'll just never get adopted into mainline. 1.4 is dead and it'll be a nice fight watching you try to keep up |
| 18:57.26 | tzafrir_laptop | drmessano, actually the right thing to ask is: "what's the overhead from this"? |
| 18:57.28 | [TK]D-Fender | Dovid: Wrong fix for your problem <------ |
| 18:57.31 | drmessano | Dovid: try throwing common sense at them, it will go MUCH farther |
| 18:58.24 | Dovid | [TK]D-Fender: What do you sugest other than loosing the GUI ? |
| 18:59.08 | [TK]D-Fender | Dovid: I suggest you look at how else you can add extensions to your system and when you need to do this. |
| 18:59.28 | drmessano | tzafrir_laptop: The right thing to ask is "Dovid, how may I help you fix a non-existant problem in Asterisk and how much will you pay me for it?" |
| 18:59.34 | drmessano | I need to be on that gravy train |
| 19:00.07 | drmessano | NAT=yes doesnt mean yes, no doesnt mean no, never means no, and always means never, and no means maybe, and yes means sometimes <---- I will fix that for you, let me show you how |
| 19:00.10 | drmessano | $$$$$ |
| 19:00.20 | tzafrir_laptop | [TK]D-Fender, the bug has a patch attached |
| 19:00.28 | Dovid | [TK]D-Fender: I actuallt spoke to a few people about the issue here and for exaclty what i needed no one had a "quick fix" |
| 19:00.59 | drmessano | WHy dont you patch the GUI |
| 19:01.08 | drmessano | Or look at the fix there |
| 19:01.31 | drmessano | If the GUI is breaking things, or not allowing a necessity, why not fix *IT* for others? |
| 19:01.41 | drmessano | Rather than a patch that will be useful in one scenario |
| 19:01.49 | [TK]D-Fender | Dovid: GUI is broken..... you don't break * just to make it MATCH. |
| 19:01.52 | Dovid | drmessano: cause as per the digium dev's it wasnt "breaking it" |
| 19:02.03 | Dovid | and it wasnt a bug. |
| 19:02.21 | [TK]D-Fender | Dovid: Yes, technically not a bug, just "not what YOU wanted". |
| 19:02.49 | Dovid | correct. so its not a gui or asterisk issue and i paid some one to "fix" it |
| 19:02.54 | drmessano | Dovid: yes, and that *ASTERISK* patch is equally useless.. If you had money to piss on something that would be ignored by the devs anyway, why pick *ASTERISK* to patch and not the *GUI* where the issue really is? |
| 19:02.58 | [TK]D-Fender | Dovid: scna for dialplan reloads and reinsert your extens with an external process. |
| 19:03.11 | tzafrir_laptop | Dovid, also: is the global context added before other includes? After other includes? |
| 19:03.12 | [TK]D-Fender | Dovid: No GUI or * code change required |
| 19:03.30 | [TK]D-Fender | tzafrir_laptop: NASTY sorting issues are bound to happen |
| 19:03.40 | [TK]D-Fender | tzafrir_laptop: this is a mistake |
| 19:03.54 | drmessano | Why not have picked the GUI to patch, and maybe the patch would be too useful to be ignored by the devs on the GUI side |
| 19:04.06 | drmessano | Instead you created something VERY likely to be ignore in *asterisk* |
| 19:04.29 | [TK]D-Fender | drmessano: Because he's clueless and throwing money at a problem. Scrthed Earth |
| 19:04.31 | Dovid | drmessano: Cause i do a lot of playing around on the side and this helped me. in general i dont use the gui |
| 19:04.34 | *** join/#asterisk dieguito84 (n=diego@87.18.187.20) |
| 19:04.36 | [TK]D-Fender | Scortched* |
| 19:04.46 | drmessano | Your left toe is too big, so you had your left eye removed so you cant see it... Fix the TOE |
| 19:05.23 | Dovid | hides in a corner |
| 19:05.24 | [TK]D-Fender | Dovid: WRONG SOLUTION |
| 19:05.44 | Dovid | [TK]D-Fender: Do you code for asterisk ? |
| 19:06.06 | [TK]D-Fender | Dovid: No, I look through it occasionally. |
| 19:06.17 | drmessano | You created a workaround in ASTERISK that is likely to never be accepted rather than patch the real problem, which is in the GUI, because "I dont use the GUI that much" |
| 19:06.20 | drmessano | God I love open source |
| 19:06.32 | Dovid | [TK]D-Fender: Ok. for future "super smart ideas" I will ask you first what the best idea is ;) |
| 19:07.07 | Dovid | drmessano: didn't think of it at the time. I was thinking about wut would help me. ME !! ME !! ME !! |
| 19:07.25 | drmessano | I guess I need to patch asterisk to listen on 5004-5082 for SIP rather than just 5060 due to users opening up a port RANGE for SIP signalling rather than config their firewall properly. |
| 19:07.32 | [TK]D-Fender | Dovid: I can come up with tons of clever hacks. I can even come up with pretty good coding ideas and where they should probably apply. I also know what is flat out cataclysmically WRONG. |
| 19:08.07 | drmessano | adds onlyworksonmybox=yes to sip.c |
| 19:08.25 | [TK]D-Fender | Dovid: And if I had C and *NIX build experience, I would be coding actively |
| 19:08.34 | drmessano | God, I wouldnt |
| 19:08.57 | drmessano | I bang my head enough hacking code together.. I dont know how these full time devs do it |
| 19:09.01 | Dovid | i would too. but i got no time to learn. time to go back to school.... |
| 19:09.26 | drmessano | I dont have the interest, so when I just need something to WORK, it pisses me off even having to screw with code |
| 19:09.44 | drmessano | Somewhat fun when it works, and works fast.. but when I get hung up on something trivial, I get stabby |
| 19:09.50 | drmessano | NO PATIENCE for coding |
| 19:10.51 | drmessano | I would like to get somewhat comfortable with PHP... just because I do a lot of web work |
| 19:15.57 | [TK]D-Fender | drmessano: PHP is the only modern-ish language I code in and even then, no OOP, etc |
| 19:16.27 | Dovid | TK: How did you learn php ? |
| 19:16.31 | [TK]D-Fender | should learn BASH & PERL |
| 19:17.21 | [TK]D-Fender | Dovid: by reading http://www.php.net/docs.php |
| 19:18.11 | seanbright | perl <3 |
| 19:18.15 | Dovid | TK: I use that, irc and lots of training vid's on line |
| 19:18.21 | [TK]D-Fender | Dovid: I understand the basics of programming through minimal OOP and have been programming all my life, just never kept up |
| 19:18.55 | [TK]D-Fender | Dovid: Syntax guide is all I need for most things. |
| 19:19.51 | [TK]D-Fender | Dovid: Syntax is what is wrong with your patch. The nature of extension sorting will screw people over. |
| 19:20.35 | Dovid | TK: Gona speak to the person that wrote it. He used to work for Digium |
| 19:20.37 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:20.42 | Dovid | thought he would be good |
| 19:21.24 | drmessano | "Used to" = Fired for excessive drinking and incompetence |
| 19:22.01 | Dovid | drmessano: do u know who i am talking about ? |
| 19:22.13 | [TK]D-Fender | Dovid: Did you NAME THEM? |
| 19:22.29 | Dovid | thought he was refrencing to some one |
| 19:22.38 | Dovid | something wrong with this: http://googlefight.com/index.php?lang=en_GB&word1=perl&word2=php |
| 19:22.38 | [TK]D-Fender | Dovid: Or are we going to play the "I know but you you don't!" game till the end of time? |
| 19:22.41 | *** join/#asterisk freddyk (n=freddy@host228-39-dynamic.50-79-r.retail.telecomitalia.it) |
| 19:23.23 | drmessano | Dovid: If it is who I think it is, I am surprised he didn't spill beer and vomit on the code. If it's not, then maybe you got some good code |
| 19:24.24 | [TK]D-Fender | Dovid: Good coders can come up with bad ideas... just look at SLA! |
| 19:24.29 | [TK]D-Fender | forgives russellb |
| 19:24.39 | drmessano | or Flash Operator Panel |
| 19:25.58 | [TK]D-Fender | * SLA = Completely wrong solution and a 2-bit hack that frustrates the majority of those just trying to set it up before realizing they're doomed to failure |
| 19:26.30 | [TK]D-Fender | (due to their circumstances not fitting the hack's scale) |
| 19:26.48 | drmessano | [TK]D-Fender: I need my new Polycom phones and my new Asterisk PBX to behave like the two-line GE Wal Mart phones we were using before. WHAT DO YOU HAVE FOR ME? |
| 19:27.11 | [TK]D-Fender | drmessano: You act like I haven't been asked that for REAL :p |
| 19:27.12 | Dovid | hehe |
| 19:27.32 | [TK]D-Fender | drmessano: I wanted to punch that guy... |
| 19:27.34 | drmessano | BUT MY GE PHONES FROM WAL MART WOULD?? WHY CANT PALYCOM AND ASTERIX? |
| 19:27.51 | Qwell | drmessano: easy. take 2 top end polycoms, connect the GE to the passthrough port on each of them |
| 19:27.57 | drmessano | Line 1, Line 2, Line 1, Line 2.. |
| 19:27.58 | Qwell | then shove the polycoms in a drawer |
| 19:28.01 | drmessano | ROFL |
| 19:28.35 | drmessano | Yeah, have Asterisk grab the line on the 4th ring |
| 19:28.43 | Dovid | TK: How much work you think it would be to have proper SLA for asterisk ? |
| 19:28.43 | drmessano | Bam, $1000 answering machine |
| 19:28.56 | drmessano | cringes |
| 19:28.58 | drmessano | Fuck SLA |
| 19:29.01 | drmessano | Pardon my french |
| 19:29.03 | [TK]D-Fender | Dovid: I think I asked oej once... not sure I got an answer |
| 19:29.10 | drmessano | Fuck *** <--- Better? |
| 19:29.44 | drmessano | People need to move past SLA.. it's 2006, people |
| 19:29.45 | Dovid | TK: He would have to re-write the Asterisk SIP stack ? |
| 19:29.52 | drmessano | Don't be all "Dave-like" about it |
| 19:29.55 | [TK]D-Fender | drmessano: * has many interfaces.... none CURRENTly suitable for your... enthusiasm ;) |
| 19:30.14 | drmessano | app_fufme? |
| 19:30.38 | [TK]D-Fender | Dovid: well DUH, its a major SIP SPEC! |
| 19:30.56 | drmessano | That would be a big seller.. The SIP SIPPER.. including a precompiled app_fufme.so |
| 19:31.00 | Dovid | ok. i remember he spoke about it. think it was called code pinaple |
| 19:31.20 | [TK]D-Fender | Dovid: No, that was just a general re-werite years ago |
| 19:31.30 | drmessano | First rule of Codename Pineapple.. DONT MENTION CODENAME PINEAPPLE IN IRC |
| 19:31.33 | drmessano | Come on, people |
| 19:31.40 | [TK]D-Fender | Dovid: and the 2 OTHER major re-writes have smoked out as well. |
| 19:32.04 | Dovid | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg162336.html |
| 19:32.08 | Dovid | so it seems |
| 19:32.12 | *** join/#asterisk ManxPower (n=manxpowe@182.sub-70-214-154.myvzw.com) |
| 19:32.13 | drmessano | Yeah, rewrite, rewrite.. Easier to get the Staple gun and hot glue |
| 19:32.25 | drmessano | Thats pretty much where TCP is |
| 19:33.04 | drmessano | picks up the phone and calls his Exchange UM.. "Hmm a fast busy".. and slams the phone down |
| 19:33.07 | drmessano | Anyway.. moving on |
| 19:36.33 | SpecialEd | GE walmart phone? |
| 19:37.50 | BadHAL | Well, I have a fairly simple question. |
| 19:38.00 | BadHAL | I have a pretty fresh * box |
| 19:38.05 | BadHAL | new to this whole thing |
| 19:38.28 | BadHAL | I am trying to find out where i set the call quality/codec |
| 19:38.44 | [TK]D-Fender | BadHAL: for sip = sip.conf |
| 19:38.52 | [TK]D-Fender | BadHAL: for iax = iax.conf |
| 19:38.57 | BadHAL | sip is what I am looking for |
| 19:39.10 | Dovid | so then sip.conf |
| 19:39.23 | BadHAL | I don't know what option(s) to set though |
| 19:39.25 | BadHAL | Is the problem |
| 19:39.40 | ManxPower | your best bet is The Asterisk book. That will help you avoid asking questions a total noob asks like you are doing now. |
| 19:39.41 | ManxPower | ~book |
| 19:39.42 | jbot | methinks book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 19:39.44 | [TK]D-Fender | BadHAL: go read the samples. you don't set "quality", the codecs themselves vary in quality and is a tradeoff against bandwidth |
| 19:40.20 | BadHAL | I have been reading the book, ill give it another look through to see if I can find the option |
| 19:40.28 | [TK]D-Fender | BadHAL: You should generally use 1 codec per peer. do "disallow=all" followed by "allow=ulaw" for example to allow only ulaw |
| 19:41.00 | BadHAL | I see |
| 19:41.08 | ManxPower | It is seldom useful to allow more than one codec and even if you do need to enable more than one codec, the codec picked is always NOT the one you wanted. Asterisk is like that. |
| 19:41.23 | BadHAL | Understood |
| 19:41.25 | [TK]D-Fender | BadHAL: start with this tidbit and go read up on the list of codec supported by your devices and calculate any bandwidth considerations. |
| 19:41.38 | BadHAL | bandwidth is not a large concern |
| 19:41.45 | BadHAL | I only have one outside SIP trunk |
| 19:41.59 | BadHAL | the rest of the network is internal and I have QoS setup |
| 19:42.05 | BadHAL | gigabit backbone |
| 19:42.28 | ManxPower | BadHAL: All that bandwidth won't help you when some router between you and your ITSP goes down |
| 19:43.12 | BadHAL | I don't forsee a router going down |
| 19:43.24 | BadHAL | this is a small network (15-20 users) |
| 19:43.27 | ManxPower | happens all the time. |
| 19:43.40 | ManxPower | BadHAL: I was not referring to a router on your network. |
| 19:43.50 | BadHAL | In that case it would not matter which codec I picked? |
| 19:43.57 | BadHAL | If a router goes down outside my network |
| 19:44.02 | BadHAL | It is going to mess things up regardless |
| 19:44.03 | ManxPower | I was referring to some random router some random provider has that your packets are going thru for some reason |
| 19:44.05 | BadHAL | assuming I am dialing out |
| 19:44.23 | BadHAL | I understand |
| 19:44.25 | ManxPower | BadHAL: My point exactly with regards to your QoS'd GigE network |
| 19:46.22 | ManxPower | I hate UPS |
| 19:46.41 | BadHAL | Enjoying your package delays? |
| 19:46.57 | BadHAL | i had 3 day shipping on an item, just recieved it yesterday |
| 19:47.02 | BadHAL | I ordered it 2 weeks ago |
| 19:47.06 | ManxPower | They returned a package to me that I sent out, said the person moved (they didn't) then claimed it was delivered -- to me. |
| 19:47.16 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 19:47.26 | ManxPower | and they see nothing wrong with that and closed my trouble ticket |
| 19:50.31 | BadHAL | Ah |
| 19:50.36 | BadHAL | GSM is the most pleasing to me |
| 19:51.57 | drmessano | Neil Diamond is the most pleasing to me |
| 19:59.51 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 20:00.04 | [TK]D-Fender | BadHAL: Internal devices should be ULAW only you |
| 20:00.27 | [TK]D-Fender | drmessano: Oh Caroline! |
| 20:01.31 | BadHAL | http://img530.imageshack.us/img530/7648/image050ny3.jpg |
| 20:01.32 | BadHAL | http://img55.imageshack.us/img55/4484/image051ml6.jpg |
| 20:01.35 | BadHAL | http://img55.imageshack.us/img55/2568/image052ui0.jpg |
| 20:01.39 | BadHAL | http://img530.imageshack.us/img530/114/image053sv5.jpg |
| 20:03.14 | MRH2 | trash can needs emptied |
| 20:03.20 | BadHAL | that's not quite trash |
| 20:03.26 | BadHAL | that's all shit that needs to be shredded |
| 20:03.39 | jaytee | Aw, Cracklin' Rosie, get on board |
| 20:03.39 | jaytee | We're gonna ride |
| 20:03.39 | jaytee | Till there ain't no more to go |
| 20:03.39 | jaytee | Taking it slow |
| 20:03.39 | jaytee | And Lord, don't you know |
| 20:03.40 | jaytee | We'll have me a time with a poor man's lady |
| 20:03.48 | BadHAL | yar emist_ |
| 20:03.48 | BadHAL | erm |
| 20:03.54 | BadHAL | wtf emailer died |
| 20:03.56 | drmessano | Red, Red, Wiiiiine.... |
| 20:03.57 | BadHAL | oops |
| 20:04.04 | BadHAL | I did that paste in the wrong channel |
| 20:04.06 | BadHAL | sorry folks |
| 20:04.11 | ManxPower | Oh look! It's a Saturday afternoon in #asterisk |
| 20:04.21 | jaytee | :-) |
| 20:04.28 | drmessano | Go to my heaaad... |
| 20:05.07 | jaytee | aw, we drove ManxPower out |
| 20:05.16 | drmessano | Make me fooooooorget that I.. still need her so... |
| 20:05.26 | [TK]D-Fender | off shopping, BBIAB |
| 20:06.03 | drmessano | It was so much quieter without him bitching endlessly about the topic of conversation for however long he was gone |
| 20:06.05 | drmessano | Not that I noticed |
| 20:06.09 | drmessano | Much.. |
| 20:06.21 | drmessano | AHEM |
| 20:06.29 | drmessano | *Inside voice* |
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| 20:09.42 | drmessano | Girl, you'll be a womaaaan.. SOON |
| 20:10.12 | drmessano | Goooogle... you'll be out of betaaaa, SOON |
| 20:11.10 | drmessano | Search you so much cant count all the ways, from sewing machines to friito lays.. |
| 20:11.14 | drmessano | Oh, shower time.. brb |
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| 21:35.25 | ziro_axis | hello all |
| 21:35.35 | ziro_axis | i have a question about *Now |
| 21:35.45 | ziro_axis | who can help |
| 21:35.51 | ziro_axis | here is the problem |
| 21:36.11 | ziro_axis | after installation of the *Now and making the configuration |
| 21:36.35 | ziro_axis | and setting the extensions |
| 21:37.09 | ziro_axis | when i want to modify or delete an extension i need to slect it. here is the point |
| 21:37.28 | *** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr) |
| 21:37.34 | ziro_axis | once i click the desired extention to modify |
| 21:37.55 | ziro_axis | all the extentions are selected |
| 21:38.21 | ziro_axis | so how can i delete extentions by comand line ?? |
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| 22:00.38 | BadHAL | Do I need to name my SIP peers a number to make phones properly detect their voicemail? |
| 22:00.52 | BadHAL | I see errors in my console when using some softphones/hardphones |
| 22:01.11 | BadHAL | 'Received SIp subscribe for peer without mailbox: blahblah.username.blah |
| 22:01.41 | BadHAL | In voicemail.conf I used the user's extension (numerical) for the mailbox |
| 22:06.03 | [TK]D-Fender | BadHAL: in sip.conf under your phones entry : mailobox=123@sectioninvoicemail.conftheyareunder |
| 22:06.42 | [TK]D-Fender | BadHAL: phones do you typically have to subscribe to voicemail, * sends out notify packets all by itself |
| 22:07.03 | BadHAL | Ahhh cool |
| 22:07.05 | BadHAL | that makes it easy |
| 22:07.11 | BadHAL | How am I missing these options? |
| 22:07.21 | BadHAL | I keep thinking I am reading the documents enough but apparently not |
| 22:07.24 | [TK]D-Fender | BadHAL: in what way? |
| 22:07.37 | [TK]D-Fender | BadHAL: and what documents? |
| 22:07.41 | BadHAL | the book |
| 22:07.56 | [TK]D-Fender | BadHAL: give the sample configs a good read over... |
| 22:08.03 | [TK]D-Fender | the Book is FAR from "complete" |
| 22:08.56 | [TK]D-Fender | BadHAL: it is good to gain an understanding of several important concepts and has a few good reference s & appendixes, but the core stuff comes in your tarball for the nitty-grity |
| 22:09.16 | BadHAL | gotcha |
| 22:11.02 | BadHAL | outstanding, worked perfectly |
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| 23:04.44 | Jabess | Some body know if the feature betwen Asterisk and Skype all voip data Is send via UDP Protocol or TCP Protocol? |
| 23:05.38 | [TK]D-Fender | Jabess: what "feature" Asterisk does not yet natively support Skype |
| 23:05.57 | Madkiss | does somebody of you know where I can configure the e-mail-adress that hylafax sends notifications to? |
| 23:06.13 | Jabess | Asterisk and Skype is nos supported? |
| 23:06.18 | Jabess | *not |
| 23:06.28 | eppigy | i think they have a channel driver |
| 23:06.31 | eppigy | module |
| 23:06.53 | [TK]D-Fender | there are THIRD PARTY options, but currently nothing officially supported |
| 23:07.08 | eppigy | yes |
| 23:07.33 | eppigy | so please direct your questions to #THIRDPARTY |
| 23:09.09 | Jabess | every word have a channel? hahaha |
| 23:09.22 | Jabess | hehehe |
| 23:09.49 | [TK]D-Fender | Jabess: yes... like "/part" for example. |
| 23:12.37 | eppigy | lol |
| 23:12.41 | Jabess | Ok, how much time could be official? |
| 23:13.12 | [TK]D-Fender | Jabess: Ask Satan when his snow-blower arrives... |
| 23:13.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 23:13.37 | eppigy | TRABAJO |
| 23:19.39 | ricko73 | Jabess: this was discussed a few weeks ago on the Voip User's Conference. (Skype for Asterisk). The person who discussed it was an official Digium representative so I'd start by listening to the archive of that podcast |
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| 23:23.12 | [TK]D-Fender | ricko73: However his answer of "when" simply isn't happening.... |
| 23:23.51 | eppigy | Jabess: early second quarter |
| 23:24.11 | Jabess | ok |
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| 23:45.39 | wwalker | anyone have experience using the different solutions for voice mail detection? |
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