| 00:00.24 | TheCop | nice setup |
| 00:01.41 | fluke | TheCop it is. I didn't get to do much except the physical install and local setup of the phones. (The phone uses ftp to fetch a config file. we did have to change a few settings for the phone to send correct dhcp options) |
| 00:02.20 | TheCop | yeah I have polycom phone |
| 00:02.25 | fluke | using codec 711 though, no 729 |
| 00:02.53 | fluke | they are nice phones. I just didn't feel like paying this much for phones for home.. |
| 00:02.53 | TheCop | I'm using 729 for Dialup VoIP station |
| 00:03.05 | *** join/#asterisk fraude (n=fraude@h8441226052.dsl.speedlinq.nl) |
| 00:03.47 | fraude | hi yall |
| 00:04.24 | *** join/#asterisk locid (n=locid@206-248-133-11.dsl.teksavvy.com) |
| 00:04.46 | fraude | NickServ REGISTER mircXS4freddy |
| 00:05.41 | fraude | l |
| 00:05.41 | fraude | l |
| 00:05.50 | fluke | fraude, missing a / there :) |
| 00:06.23 | fraude | :) |
| 00:06.31 | fraude | damn.. |
| 00:07.18 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 00:07.28 | _Thor | Hello everyone |
| 00:08.47 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 00:09.10 | *** join/#asterisk xbmodder (i=nobody@unaffiliated/xbmodder) |
| 00:09.17 | xbmodder | why isn't my sipura registering? |
| 00:09.24 | xbmodder | sip show registry |
| 00:09.24 | xbmodder | Host Username Refresh State |
| 00:09.24 | xbmodder | voip-co3.teliax.com:5060 xbmodder 105 Registered |
| 00:09.56 | ast_freak | xbmodder, sip debug |
| 00:10.45 | fraude | im quite new to this asterisk - PBX .. would you advise me to get meself a digium (staterskit) to connect thru a POTS-line? |
| 00:10.48 | xbmodder | I see a bunch of registers |
| 00:11.16 | xbmodder | ast_freak, this is what happens on my sipura (after one digit) pauses and starts to BEEP |
| 00:11.27 | fluke | fraude funny you ask this, I was asking myself the same question about 30 minutes ago (well, I've looked at it multiple times but still don't know what hardware to start with) |
| 00:11.42 | xbmodder | Jan 2 17:07:00 10.0.0.210 SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.0.0.210:5060;branch=z9hG4bK-2ae259e4;received=10.0.0.210^M From: sipura <sip:sipura@10.0.0.1>;tag=9512537514b7287o0^M To: sipura <sip:sipura@10.0.0.1>;tag=as7670afae^M Call-ID: a595f5ad-565575af@10.0.0.210^M CSeq: 22 REGISTER^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Max-Forwards: 70^M Expires: 60^M Contact: <sip:sipura@10.0.0 |
| 00:11.42 | xbmodder | .210:5060>;expires=60^M Date: Tue, 03 Jan 2006 00:11:13 GMT^M Content-Length: 0 |
| 00:11.51 | xbmodder | last peice of info I get |
| 00:11.58 | xbmodder | 10.0.0.210 is sipura |
| 00:12.42 | fraude | it's a bit foggy, isn't it. I would like to make a test-PBX.. but don't know how to exactly start :( |
| 00:12.58 | SkramX | What are you trying to do? |
| 00:13.25 | fraude | i would like to connect to some SIP-clients at 1st. |
| 00:13.48 | fraude | 2nd I would like to connect to normal POTS-subscribers |
| 00:14.13 | fluke | fraude have you got a linux (or other supported unix-like system) to install asterisk on? you could start with a soft phone on either a windows or mac or linux machine |
| 00:14.59 | xbmodder | ast_freak, it can recieve calls (I called it from console) but it can't put out calls :-| |
| 00:14.59 | Kumbang | guys, what is the debian package for svn |
| 00:15.01 | fluke | I mean, a machine that will be the linux _server_ and then a soft phone on a win/mac/linux _station_ |
| 00:15.38 | fluke | kumbang I haven't been on debian in a while but I guess apt-cache search subversion shall give you some good results |
| 00:16.03 | fraude | use aptitude |
| 00:16.14 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 00:16.22 | Kumbang | ok thanks |
| 00:16.26 | xbmodder | Does anyone here know how to make it work? |
| 00:16.48 | ast_freak | xbmodder, what does your sip.conf entry for the sipura look like? |
| 00:17.48 | xbmodder | http://pastebin.com/487906 |
| 00:18.30 | xbmodder | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) < my sipura Dial Plan |
| 00:18.52 | xbmodder | ast_freak, are the digits send to the PBX upon dialing |
| 00:18.57 | xbmodder | or does the ATA manage that? |
| 00:19.11 | ast_freak | Did you set the authuser in the sipura config? |
| 00:19.47 | fraude | i'd like to get a good asterisk tutorial .. B4 asking stupid questions in this channel.. Can anyone help? |
| 00:20.11 | ast_freak | ~wiki |
| 00:20.31 | fraude | ? |
| 00:20.55 | xbmodder | http://xbmodder.us/tmp/sipura.pdf |
| 00:20.55 | ast_freak | http://voip-info.org/ |
| 00:21.01 | xbmodder | ast_freak, http://xbmodder.us/tmp/sipura.pdf < my sipura settings |
| 00:22.07 | xbmodder | fraude, look at examples and voip-info.org |
| 00:22.15 | fluke | fraude there's asteriskdocs.org and voip-info.org |
| 00:22.30 | xbmodder | I have sucked down over 50MB of docs from voip-infp |
| 00:23.18 | ast_freak | xbmodder, sorry, I can't view your PDF quite right. |
| 00:23.56 | _Thor | hi, anyone knows how to unlock a dta-310? |
| 00:24.33 | xbmodder | http://xbmodder.us/tmp/sipura-0.jpg and http://xbmodder.us/tmp/sipura-1.jpg |
| 00:24.36 | fraude | great URL's folks.. thnx |
| 00:25.40 | ast_freak | xbmodder, in sip.conf, username=sipura is your Auth ID in sipura config. |
| 00:26.02 | ast_freak | Or you could just remove username=sipura from the sip.conf |
| 00:26.23 | xbmodder | ok. |
| 00:26.39 | ast_freak | then try it |
| 00:26.56 | fraude | NL - people here? |
| 00:27.03 | *** join/#asterisk freezer (i=leetiden@ACB4BC3A.ipt.aol.com) |
| 00:30.51 | xbmodder | nope |
| 00:31.22 | *** join/#asterisk tina (n=tina@viper.ouraynet.com) |
| 00:31.30 | _Thor | anybody knows how to unlock a dta-310? |
| 00:31.56 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 00:32.23 | xbmodder | ast_freak, nope |
| 00:32.32 | nick125 | hey, i got a quick question: I'm looking for a provider that will terminate calls to the US/48 that will do it for under 1 cent a minute USD, any ideas? |
| 00:32.39 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:33.50 | wunderkin | i forgot the name of a company i knew of out here.. havent used them yet but i met the ceo |
| 00:34.07 | wunderkin | they are a couple racks next to me |
| 00:34.49 | _Thor | I will take this silence as a no |
| 00:36.21 | xbmodder | _Thor, sure |
| 00:36.40 | _Thor | Thanks a lot |
| 00:37.00 | xbmodder | _Thor, I mean sure "take this silence as a no" |
| 00:37.09 | _Thor | I know |
| 00:37.42 | _Thor | I mean thanks a lot everybody |
| 00:38.04 | xbmodder | _Thor, welcome |
| 00:38.09 | ast_freak | xbmodder, what does your sip debug say now. Do you have verbose set at 4 |
| 00:38.10 | ast_freak | ? |
| 00:38.53 | *** join/#asterisk tina_ (n=tina@viper.ouraynet.com) |
| 00:40.50 | xbmodder | ast_freak, it doesn't give any errors |
| 00:41.20 | xbmodder | Jan 2 17:36:05 10.0.0.210 SIP/2.0 401 Unauthorized^M Via: SIP/2.0/UDP 10.0.0.210:5060;branch=z9hG4bK-e1a04914; |
| 00:41.20 | xbmodder | received=10.0.0.210^M From: sipura <sip:sipura@10.0.0.1>;tag=fbc93c0d89a3e2cco0^M To: sipura <sip:sipura@10.0.0 |
| 00:41.20 | xbmodder | .1>;tag=as6f6b071a^M Call-ID: 949b8bd1-3a3e35d8@10.0.0.210^M CSeq: 21 REGISTER^M User-Agent: Asterisk PBX^M All |
| 00:41.20 | xbmodder | ow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Max-Forwards: 70^M Contact: <sip:sipura@10.0. |
| 00:41.22 | xbmodder | 0.1>^M WWW-Authenticate: Digest realm="asterisk", nonce="75d05b9f"^M Content-Length: 0^M ^M |
| 00:41.26 | xbmodder | my sipura says that |
| 00:43.19 | fraude | the O'reily "The future of telefony" PDF IS GREAT |
| 00:43.39 | fraude | the foreword makes me very curious of the rest. |
| 00:43.57 | implicit | only for those who are naieve enough to believe it |
| 00:44.19 | JunK-Y | hey implicit, sup man? |
| 00:44.27 | fraude | i've invented naievety |
| 00:44.30 | Ariel_ | xbmodder, it says unauthorized |
| 00:44.35 | implicit | sup JunK-Y ! |
| 00:44.37 | implicit | howa reu |
| 00:44.37 | implicit | :) |
| 00:44.44 | implicit | ;) |
| 00:44.44 | xbmodder | Ariel_, yeah, I noticed that afterwards |
| 00:44.45 | *** join/#asterisk classicx (n=classic_@gb.jb.102.37.revip.asianet.co.th) |
| 00:44.52 | implicit | fraude, i'm just kidding with you though |
| 00:44.55 | implicit | havn't seen it |
| 00:45.31 | Ariel_ | hello JunK-Y hope all is well thing new year.... 2006 wow |
| 00:45.49 | JunK-Y | ya, happy new year ariel, have a good one. |
| 00:45.49 | fraude | this foreword is the evolution of Asterisk (even Linux) in a NUTS-shell.. |
| 00:46.10 | fraude | i'll proceed reading.. see ya'll later |
| 00:47.17 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-70-93-239-175.san.res.rr.com) |
| 00:50.35 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:51.44 | ravenpi | I saw some stuff a few minutes ago about people looking to start. I won't try to point them toward hardware, but let me just say that the new O'Reilly Asterisk book explains things in a much more... cohesive way than the other available docs. Super handy, and definitely worth the $$$ -- esp. since it can be had for free at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 ... |
| 00:51.46 | ravenpi | ...(though I did buy it). |
| 00:52.05 | *** join/#asterisk tetsuzan (n=rider@201.2.206.4) |
| 00:52.47 | shmaltz | ~seen bweschke |
| 00:52.50 | jbot | bweschke is currently on #asterisk (1h 25m 17s) |
| 00:59.57 | *** join/#asterisk anonymouz666 (n=lynx@ns2.redetaho.com.br) |
| 01:00.01 | *** join/#asterisk tina (n=tina@viper.ouraynet.com) |
| 01:00.31 | anonymouz666 | It takes a while to learn all Asterisk apps |
| 01:00.36 | anonymouz666 | I must have patience |
| 01:03.18 | *** join/#asterisk _cleric_ (n=dacleric@p5482ACEC.dip0.t-ipconnect.de) |
| 01:04.01 | fluke | ravenpi thanks for the info. I happen to have found the pdf about an hour ago and I also pointed the other beginner to the same page.. thanks a bunch! (I'll probably end up buying it too.. I own about 30 or 40 oreilly books....) |
| 01:04.22 | anonymouz666 | Asterisk - The Future of Telephony? |
| 01:04.25 | anonymouz666 | Very nice book |
| 01:04.50 | Kumbang | doh, still d-channel wont bring to came UP |
| 01:05.36 | Kumbang | span=2,0,0,ccs,hdb3,crc4 |
| 01:05.53 | Kumbang | not work for te110p |
| 01:06.02 | ravenpi | It really is. I've been doing telecom and Linux (but not together) for over ten years -- and just couldn't find docs to explain stooopid Asterisk-specific stuff to me. A-TFoT, however, does a really good job. *no longer feels entirely lame* |
| 01:07.02 | brockj49464 | Any idea how to solve "Got SIP response 423 "Interval Too Brief" back from 69.25.48.85" with ZingoTel? |
| 01:08.09 | *** join/#asterisk Weezey (n=ohno@206.210.109.226) |
| 01:08.28 | Weezey | why is SetAccount not a built-in function anymore? |
| 01:08.52 | *** join/#asterisk OloBola (n=not@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
| 01:08.55 | *** join/#asterisk GXTi (i=realme@freenode/developer/GXTi) |
| 01:09.32 | wunderkin | you mean not an application, it is a function now |
| 01:09.59 | *** join/#asterisk nereayfran82 (n=ircap8@24.Red-83-51-237.dynamicIP.rima-tde.net) |
| 01:10.43 | Weezey | an older cvs has SetAccount |
| 01:10.52 | Weezey | my svn doesn't have it. |
| 01:11.13 | wunderkin | right.. it was deprecated |
| 01:11.35 | Weezey | what should I be using now? |
| 01:12.46 | OloBola | any suggestions on how to get Fedora 3 to recognize my x100p clone (motorola 62802)? On boot it recognized new hardware and installed it as an unknown modem |
| 01:13.19 | TheCop | What's the most used echo cancellation module in zaptel for Quebec, Canada ? |
| 01:13.24 | ast_freak | xbmodder, does your console say that the sipura is registering yet? |
| 01:13.44 | wunderkin | Weezey, SetAccount has been deprecated in favor of the Set(CDR(accountcode)=account). |
| 01:14.34 | Weezey | awesome, thanks. |
| 01:14.36 | fluke | TheCop I wish I knew too, I'm in Montreal.. |
| 01:15.06 | *** join/#asterisk Manolo (n=peroo@200.124.172.72) |
| 01:15.27 | TheCop | fluke, I'm using mark the default |
| 01:15.33 | TheCop | and I have some sometimes |
| 01:15.44 | *** join/#asterisk nose2 (n=ircap8@63.245.87.169) |
| 01:15.51 | *** join/#asterisk Thazza (n=me@203.80.44.200) |
| 01:16.08 | nose2 | hi |
| 01:16.11 | fluke | we've got echo at work, mostly calling 514 or 450, not long distance. afaik, we're using the cards without hardware echo cancel, and will probably be changing the cards soon... |
| 01:16.13 | Thazza | Hey all |
| 01:16.19 | Weezey | wunderkin: I'm gonna add that to the wiki. |
| 01:16.37 | wunderkin | yes its not on there yet |
| 01:16.40 | *** join/#asterisk Cucurucho (n=peroo@200.124.172.72) |
| 01:16.54 | nose2 | i am new with asterkisk i want know wath is the system requeriments |
| 01:16.57 | Thazza | I have a problem, was wondering if someone could help.. I am having issues transfering calls i have made.. Yet incomming calls i can transfer to another device. |
| 01:16.59 | TheCop | fluke, tdm400 or PRI ? |
| 01:17.28 | fluke | TheCop: definitely all PRIs. not sure which card exactly :-/ |
| 01:17.29 | ast_freak | Thazza look at your options for Dial() |
| 01:18.01 | Thazza | Where would i find that ast_freak? |
| 01:18.13 | ast_freak | nose2: Depends on what you want to do. Bottom line would be > 233 Mhz, 128 RAM, etc. |
| 01:18.24 | Cucurucho | can anyone help me here: i want to direct all incoming calls from PSTN to the three FXO lines i got. I only have one working. so when there´s a call and the line is occupied, it does not transfer to the next open line, but it gets the busy tone instead |
| 01:19.03 | ast_freak | Thazza, 'show application dial' in the console. |
| 01:19.20 | *** join/#asterisk woodchuck (n=woodchuc@S0106000000da2a3d.ok.shawcable.net) |
| 01:19.29 | nose2 | thanks ast_freak asterisk work better in freebsd or linux? |
| 01:19.34 | Thazza | cool. thanks ast_freak. |
| 01:19.48 | ast_freak | Cucurucho, you need to talk to your PSTN provider about that. They set up the rollover on their end. |
| 01:19.53 | ast_freak | Thazza, no prob. |
| 01:20.12 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:20.15 | Thazza | ast_freak. Where are the transfer options in the asterisk config files? stored. |
| 01:20.26 | ast_freak | nose2, I prefer Slackware linux, but other *nix works just as well from what I've heard. |
| 01:20.59 | ast_freak | Thazza, extensions.conf -- find the part of the dialplan that does the dialing, and change the option. |
| 01:21.02 | Cucurucho | ast_freak, no man there´s not the issue.. i just want to know how to config my asterisk, so that it takes incoming calls to the three extensions i want |
| 01:21.04 | nose2 | thanks ast_freak :) |
| 01:21.10 | fluke | nose2, ast_freak: I guess it mostly depends on whether you use sip/iax only, or if you have pci cards.. (driver availability?) |
| 01:21.38 | Thazza | ast_freak: This is for outgoing dialing as well? |
| 01:22.25 | Cucurucho | I want to take my incoming calls from PSTN to three different extensions, so that no one that calls gets the busy tone. |
| 01:23.27 | ast_freak | Boy, you guys are working me hard today :) |
| 01:23.57 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 01:23.59 | Thazza | ast_freak.. Well when you are good.. You are worked hard. ;-) |
| 01:24.41 | ast_freak | Cucurucho, check out how dial() works 'show application dial' - or - check out queues |
| 01:24.50 | ast_freak | nose2, You're welcome. |
| 01:24.52 | Cucurucho | Hellooo any help with that please? |
| 01:25.31 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 01:26.05 | ast_freak | Thazza, extensions.conf handles all the dialplan unless you have something set up in extensions.ael |
| 01:27.01 | Thazza | ast_freak. Just out of interest.. (and yes i know it is junk) yet you wouldn't know where the transfer section would be in AMP. |
| 01:27.17 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 01:28.16 | ast_freak | Sorry, can't help you with AMP or AAH. I've never used them. |
| 01:28.52 | ast_freak | Cucurucho, did you figure it out? |
| 01:28.58 | Cucurucho | One more thing.. how can i config, what to dial, when i want to take a call from an extension different for whats ringing |
| 01:29.32 | fluke | thanks everyone, good evening. going out for dinner! |
| 01:30.24 | Cucurucho | with "show app dial" you mean? |
| 01:30.36 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:30.42 | ast_freak | Cucurucho, I'm not sure I understood your question. |
| 01:30.59 | *** join/#asterisk linagee (n=linagee@netblock-68-183-1-214.dslextreme.com) |
| 01:31.21 | Cucurucho | ast_freak, me neither |
| 01:32.05 | ast_freak | You want calls to go to ext1, then ext2 if ext1 is busy, then ext3, etc.? |
| 01:32.18 | Cucurucho | ast_freak, EXACTLY |
| 01:33.20 | _Sam-- | could use DIALSTATUS? |
| 01:33.56 | ast_freak | Cucurucho, look at how dial works. type 'show applicaton dial' in the console. You can setup the dialplan to dial ext1, and if busy, it will go to a different priority. |
| 01:34.52 | ast_freak | Cucurucho, -- or -- you can set up a queue and put the extensions into the queue. |
| 01:35.26 | Cucurucho | ast_freak, i can setp the dialplan from what .conf file?? (THnx for answering all this noob shit) |
| 01:36.12 | ast_freak | Cucurucho, extensions.conf |
| 01:36.16 | *** join/#asterisk david-c (n=dcoulson@207.166.203.178) |
| 01:39.04 | Cucurucho | ast_freak, thnx |
| 01:40.16 | hackeron | hey, I'm trying to use DISA and I'm getting a dialtone, but as soon as I dial the first 2 numberso f the extension, asterisk just hangs up, any ideas? |
| 01:40.27 | *** join/#asterisk JunK-Y_ (n=junky@69.156.218.24) |
| 01:40.36 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:40.57 | *** join/#asterisk Cinen (n=Cinen@cpe-065-188-184-160.triad.res.rr.com) |
| 01:43.48 | *** part/#asterisk Cinen (n=Cinen@cpe-065-188-184-160.triad.res.rr.com) |
| 01:43.52 | *** join/#asterisk Cinen (n=Cinen@cpe-065-188-184-160.triad.res.rr.com) |
| 01:44.05 | *** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
| 01:45.19 | Lloydie-t | Hi I need some help trying to get realtime 'sippeers to work' |
| 01:45.32 | Lloydie-t | I get this error 'Realtime mapping for 'sippeers' found to engine 'res_sqlite', but the engine is not available' |
| 01:45.50 | MikeJ[Laptop] | hmmm |
| 01:45.55 | MikeJ[Laptop] | is sqlite running? |
| 01:46.50 | Lloydie-t | I thought the library for sqlite was built into res_sqlite?? |
| 01:46.59 | MikeJ[Laptop] | ummm |
| 01:47.04 | MikeJ[Laptop] | it may be |
| 01:47.13 | MikeJ[Laptop] | hmmm |
| 01:47.21 | NewSole | but does it not need a server to connect to |
| 01:47.33 | MikeJ[Laptop] | it may not... |
| 01:47.37 | Lloydie-t | No |
| 01:47.39 | MikeJ[Laptop] | you know what, he is right.. |
| 01:47.45 | MikeJ[Laptop] | it opens the db directly.. |
| 01:48.00 | MikeJ[Laptop] | pointed to a real db, tables set up? |
| 01:48.15 | Lloydie-t | I may be using the wrong engine name |
| 01:48.41 | MikeJ[Laptop] | I don't know res_sqlite well.. |
| 01:48.51 | MikeJ[Laptop] | sorry :( |
| 01:49.24 | NewSole | Dam I wish these spam people would get thing right..... |
| 01:49.41 | Lloydie-t | No problem, I'll google |
| 01:49.54 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 01:50.15 | NewSole | the Girlfriend keeps getting Penis Enlarment pills and I get online dating..... |
| 01:50.45 | *** join/#asterisk Los415 (n=los415@c-24-126-63-65.hsd1.ca.comcast.net) |
| 01:50.51 | _Thor | hackeron: codec problem, also... set dtmf to rfc2833 |
| 01:51.56 | Los415 | hey does anyone know how to extend the dialtime on a zap port |
| 01:52.23 | Los415 | so it dosnt go to busy signal so fast if they dont hit the #'s fast enough |
| 01:52.25 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 01:52.26 | shmaltz | Los415, Dial(zap/g1/numbwer,puttimehere) |
| 01:52.55 | shmaltz | Los415, Dial(zap/g1/wwnumber) |
| 01:52.57 | shmaltz | each w gives a 500 ms pause |
| 01:53.12 | hackeron | _Thor: codec problem? -- It says requested ulaw accepted ulaw -- no transcoding errors what so ever |
| 01:53.30 | _Thor | Newsole: actually, I've read from reliable sources that the penis indeed can be enlarged |
| 01:53.33 | hackeron | _Thor: and I'm running asterisk -cvvvvv which is not reporting any errors |
| 01:53.34 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net) |
| 01:54.06 | fugitivo | _Thor: spam? |
| 01:55.06 | _Thor | hackeron: sorry, I have the exact same problem you have but I haven't been able to solve it... all I know is that it happens to me when I am using IAX, when using SIP, I don't have that problem |
| 01:55.22 | _Thor | Fugitivo: no spam, real reliable source |
| 01:55.32 | hackeron | _Thor: ah, ok, I'll try using sip, thanks |
| 01:55.56 | fugitivo | isn't spam a real reliable source? |
| 01:56.07 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:56.35 | _Thor | hackeron: although it still beats the heck out of me why it doesn't work on IAX. btw, ulaw/alaw problem is a codec problem |
| 01:57.35 | hackeron | _Thor: what ulaw/alaw problem? |
| 01:57.49 | _Thor | fugitivo: see my friend... it's a muscle right?, so you can grow any muscle, right?... all you need is to put it to do some weight lifting <g> <g> |
| 01:59.09 | fugitivo | that remembers me a movie i saw a while ago |
| 01:59.33 | [av]bani | hm, can spa-3000 be configured by dhcp like hardphones can? |
| 02:00.02 | _Thor | hackeron: sorry, I forgot, but a few weeks ago when I was working on it, it had everything to do with the codec |
| 02:00.04 | _Sam-- | [av] you test any external FXO gateways yet? interested to know what works |
| 02:00.09 | [av]bani | spa-3000 |
| 02:00.10 | [av]bani | works |
| 02:00.43 | hackeron | _Thor: so what do you recommend I use? |
| 02:01.48 | *** join/#asterisk _Soul_ (n=Soul@87-196-34-150.net.novis.pt) |
| 02:02.02 | _Sam-- | has anyone seen the 'knopsterisk' distribution? (knoppix/asterisk) |
| 02:02.07 | _Soul_ | greetings |
| 02:02.38 | _Soul_ | just received a cisco 7940, and i'd like to try it out with our asterisk |
| 02:02.58 | _Sam-- | go for it |
| 02:03.12 | _Soul_ | i have theses files: |
| 02:03.12 | _Soul_ | 29-12-2005 16:06 14 OS79XX.TXT |
| 02:03.13 | _Soul_ | 29-12-2005 16:06 486,570 P0S3-06-2-00.bin |
| 02:03.13 | _Soul_ | 29-12-2005 16:06 486,974 P0S3-06-2-00.sbn |
| 02:03.13 | _Soul_ | 03-01-2006 01:44 1,454 SIPDefault.cnf |
| 02:03.34 | _Soul_ | i think i configured the SipDefault.cnf well, and i was trying to serve them using tftp |
| 02:03.52 | _Soul_ | the 7940 receives its ip address, and asks for the files. but then... : |
| 02:04.21 | _Thor | hackeron: use sip... I have to get it to work on iax asap, as soon as I do, I will send you a note |
| 02:04.22 | hackeron | what is required for WaitExten to accept a particular extension? -- It works for the default 1234 but not extensions I added. Any ideas? |
| 02:04.42 | hackeron | _Thor: thanks, its not urgent or anything, but thanks a bunch. |
| 02:04.44 | _Soul_ | read request for OS79XX.TXT, ok |
| 02:05.15 | _Soul_ | read request for POS3-06-.bin, failed |
| 02:06.18 | _Soul_ | inside OS79XX.TXT i have just "P0S3-06-2-00", so why is he fetching POS3-06-.bin instead of POS3-06-2-00.bin ? is this a known bug ? |
| 02:06.22 | slappingt | hey guys, I have a Sipura 300 with my asterisk set up and I am getting alot of dropped calls FXS/FXO. Any ideas on what to check? |
| 02:06.57 | _Soul_ | and if i rename POS3-06-2-00.bin to POS3-06-.bin, i get a "invalid file" |
| 02:07.08 | _Soul_ | is this a known bug ? |
| 02:09.15 | *** join/#asterisk vmlinuz (n=nabudoco@ns1.ensenada.gob.mx) |
| 02:09.24 | _DAW-LAPTOP | hey everone |
| 02:12.15 | Qwell | _Soul_: You need to rename the file, AND change what is in the config |
| 02:12.38 | Qwell | _Soul_: So, for instance, in the case above, I'd do P0S30620.bin |
| 02:12.49 | linlin | yay I won my X100P FXO auction :) |
| 02:12.51 | Qwell | and change what's in the config to P0S30620 |
| 02:13.07 | Qwell | linlin: I wouldn't call that winning |
| 02:13.41 | linlin | why? shitty card i hear |
| 02:13.44 | Qwell | indeed |
| 02:13.53 | linlin | ok for single line home use though right? |
| 02:13.58 | *** join/#asterisk SLiCKFX (n=askme@pcp03218165pcs.hlcrs201.al.comcast.net) |
| 02:14.06 | _Soul_ | Qwell, ok, i changed the OS79XX.TXT contents to P0S306200, lets try |
| 02:14.29 | Lloydie-t | in the addons there is a directory to make the res_sqlite3, but it's failing |
| 02:14.37 | Qwell | _Soul_: It needs to be 8.3 format |
| 02:14.54 | Lloydie-t | with '/bin/sh: line 1: /usr/src/asterisk/contrib/scripts/astxs: No such file or directory' |
| 02:14.57 | Qwell | P0S306200 is 9 |
| 02:15.09 | Lloydie-t | Can I do anything about it? |
| 02:15.33 | Qwell | Lloydie-t: needs to grab astxs I guess.. |
| 02:15.38 | Qwell | drop it in that dir? |
| 02:15.43 | *** join/#asterisk [hC] (n=lisa@209.200.137.24) |
| 02:15.47 | Qwell | [hC]: y0 |
| 02:15.54 | [hC] | sup qwell |
| 02:16.05 | [hC] | <- in costa rica now setting up a pbx :) |
| 02:16.06 | _Soul_ | Qwell, renamed, lets see.. |
| 02:16.10 | Qwell | [hC]: bastard |
| 02:16.32 | _Soul_ | Qwell, if that works im gonna shoot myself in the head |
| 02:16.33 | [hC] | :P |
| 02:16.38 | Qwell | _Soul_: don't do that... |
| 02:16.40 | *** part/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 02:16.42 | Qwell | _Soul_: Just paypal me instead. D: |
| 02:16.56 | _Soul_ | and then shoot update voip-info.org ;) |
| 02:17.05 | [hC] | whatcha trying to get working? |
| 02:17.12 | Qwell | [hC]: 7960 firmware... |
| 02:17.18 | [hC] | ahh.. thats fun |
| 02:17.27 | [hC] | i think ive seen just about every error message so far |
| 02:17.28 | Qwell | after you've done it a few times, you know all the quirks, heh |
| 02:17.30 | _Soul_ | Qwell, OS79XX.TXT requested ok |
| 02:17.32 | [hC] | and every workaround |
| 02:17.33 | Qwell | exactly |
| 02:17.39 | [hC] | :P |
| 02:17.51 | Qwell | _Soul_: now it should grab that file, and all the lights will start flashing |
| 02:17.53 | [hC] | one fun thing, OS79XX.TXT is finicky about newline types |
| 02:17.59 | *** join/#asterisk Thazza (n=me@203.80.44.200) |
| 02:18.00 | _Soul_ | then: read request for file POS30620.bin: PEER RETURNS ERROR, aborting transfer |
| 02:18.30 | [hC] | you know its supposed to be P zero not P 'o' right? |
| 02:18.39 | Lloydie-t | I my problem is that I built my asterisk in 'asterisk-1.2', which my be why I can't build the res_sqlite |
| 02:18.40 | Qwell | yeah...that matters a great deal, heh |
| 02:18.40 | _Soul_ | so the renaming worked, but now it does not like POS30620.bin |
| 02:18.45 | Qwell | P0S, not POS |
| 02:18.51 | Qwell | ie; it isn't a piece of shit :) |
| 02:18.56 | [hC] | heheh |
| 02:18.57 | Qwell | (unless you spell it wrong) |
| 02:19.09 | _Soul_ | pasted from the file itself: P0S30620 |
| 02:19.23 | [hC] | ah so you just retyped it that last time |
| 02:19.27 | [hC] | cause that had an O in it |
| 02:19.32 | _Soul_ | yes, my bad |
| 02:19.53 | Qwell | not sure why it would be aborting it... |
| 02:20.00 | Qwell | What firmware are you coming from? |
| 02:20.03 | [hC] | so your tftp server says that peer aborted,. or your phone? |
| 02:20.14 | _Soul_ | tftp server says that |
| 02:20.19 | Qwell | try it again.. |
| 02:20.23 | _Soul_ | the phone only says upgrading software |
| 02:20.47 | [hC] | what tftp server are you using? |
| 02:20.54 | [hC] | path/permissions correct? |
| 02:21.07 | Qwell | [hC]: it's able to grab the OS79xx.txt just fine |
| 02:21.16 | [hC] | perms then? :) |
| 02:21.22 | _Soul_ | tftpd32, unfortunatly. i«m much used to the tftpd on linux |
| 02:21.23 | Qwell | perhaps, on the .bin |
| 02:21.30 | [hC] | case sensitivity |
| 02:21.54 | [hC] | i guess if its on windows it wouldnt matter. |
| 02:21.54 | Qwell | You *DID* also rename the .sbn too, right? :) |
| 02:22.08 | [hC] | and the .loads |
| 02:22.09 | _Soul_ | running with no security, the phone can access everything in the folder |
| 02:22.17 | [hC] | but the tftp server gives an access error, |
| 02:22.25 | _Soul_ | i have no .loads, and the .sbn is renamed |
| 02:22.27 | Qwell | _Soul_: do you even have the .loans? |
| 02:22.29 | [hC] | unfortunately im not familiar with windows tftp server error messages |
| 02:22.30 | Qwell | odd |
| 02:22.36 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 02:22.40 | _Soul_ | i have no .loads |
| 02:22.46 | [hC] | actually i dont think you NEED .loads |
| 02:22.51 | Qwell | Why are you using 6.2 anyhow? Isn't it up to like 7.3? |
| 02:22.55 | Qwell | or is that just sccp? |
| 02:23.00 | [hC] | its up there |
| 02:23.09 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 02:23.12 | [hC] | but i know for sure that one of the 7.x releases (not sure if its 7.2 or 7.3) has weird issues |
| 02:23.16 | [hC] | with audio dropouts |
| 02:23.20 | [hC] | took me WEEKS to solve that. |
| 02:23.20 | Qwell | They *ALL* have weird issues. ;] |
| 02:23.39 | [hC] | haha |
| 02:23.43 | Qwell | 7970 had a problem with getting caught in a reboot loop...that was uncool |
| 02:23.44 | _Soul_ | i bought this refurbished phone, and a friend who has about 5 phones sent me these files. i dont even have an sccp license, the phone came with no licence |
| 02:24.20 | Qwell | what firmware is on it now? |
| 02:24.24 | [hC] | Qwell: i had that on mine too. forget what fixed it, but i got it eventually. |
| 02:24.27 | _Soul_ | how can i tell ? |
| 02:24.35 | Qwell | check in settings |
| 02:24.36 | [hC] | in system info after the phone boots |
| 02:24.45 | _Soul_ | the phone does not boot ;) |
| 02:24.51 | Qwell | so...none :p |
| 02:24.54 | _Soul_ | i came with no license |
| 02:24.55 | slappingt | is there a log file in asterisk that shows dropped call details? |
| 02:25.05 | Lloydie-t | I had a look in my 'usr/src/asterisk-1.2/contrib/scripts' and the file astxs. |
| 02:25.47 | Lloydie-t | Am I going to have to rebuildd the whole thing in 'usr/src/asterisk' to get res_sqlite3 to work? |
| 02:26.26 | Lloydie-t | or maybe some clever --prefix? |
| 02:26.34 | _Soul_ | i read somewhere that one needs to upgrade these phone to a certain firmware, before upgrading to the next one, is this true ? |
| 02:26.45 | Qwell | Lloydie-t: Just change the ASTDIR in the Makefile |
| 02:26.47 | Qwell | _Soul_: usually |
| 02:27.01 | [hC] | possibly, but the firmware you're using is fairly versatile from what ive seen so far |
| 02:27.13 | _Soul_ | can the 6.2.0 not load because i dont have a sufficiently recent firmware ? |
| 02:27.37 | _Soul_ | i dont know what to do, any suggestions ? |
| 02:27.54 | [hC] | unfortunately its difficult to debug those tftp messages because i dont know what they mean |
| 02:28.07 | [hC] | if you were using atftpd i would know |
| 02:28.14 | Lloydie-t | hmmm, Qwell I'll give it a try. I no linux guru. |
| 02:28.16 | [hC] | it sounds though like its having a hard time finding the image you're asking for |
| 02:28.38 | [hC] | whats inside OS79XX.txt? |
| 02:28.47 | [hC] | paste it if you can |
| 02:30.07 | _Soul_ | P0S30620 |
| 02:30.59 | [hC] | so then youve also got P0S30620.bin and P0S30620.sbn in that same directory |
| 02:31.06 | _Soul_ | just noticed that OS79XX.txt is dos formated, gonna try to convert it to unix format |
| 02:31.13 | [hC] | er, .sb2 |
| 02:31.21 | _Soul_ | [hC], yes |
| 02:31.26 | Qwell | .sbn |
| 02:31.30 | [hC] | i forget which is which |
| 02:31.31 | [hC] | :) |
| 02:31.31 | Qwell | is what he has, anyhow |
| 02:31.32 | _Soul_ | [hC], i have .sbn, not .sb2 |
| 02:32.00 | _Soul_ | i think i can paste here the tftpd log, wait |
| 02:32.04 | Qwell | ~pb |
| 02:32.09 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:32.18 | *** join/#asterisk lrizzo (n=luigi@host114-164.pool8259.interbusiness.it) |
| 02:32.32 | Qwell | lrizzo: y0 |
| 02:32.46 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 02:33.28 | [hC] | hrmm. i wonder if ive got this zapata.conf right |
| 02:33.43 | Lloydie-t | more problems /usr/src/asterisk-1.2/contrib/scripts/astxs: Permission denied |
| 02:33.48 | _Soul_ | http://pastebin.com/488090 |
| 02:34.01 | Qwell | Lloydie-t: are you root? |
| 02:34.17 | Lloydie-t | I logged in as root |
| 02:34.52 | Lloydie-t | BTW that was make that brought that error |
| 02:35.01 | _Soul_ | wait.. here's some update.. |
| 02:35.04 | Qwell | _Soul_: tried another tftpd? |
| 02:35.14 | _Soul_ | i changed the tftpd server, and now i can see: |
| 02:35.31 | [hC] | yeah that tftpd server seems bunk, the error is empty |
| 02:35.36 | *** join/#asterisk cripito (n=ncripito@c-67-161-130-59.hsd1.co.comcast.net) |
| 02:35.40 | cripito | hi |
| 02:35.44 | _Soul_ | the file is transmited till about 80%, and the phone resets the transmission |
| 02:35.51 | Lloydie-t | Shall I change the permissions |
| 02:35.54 | Qwell | too large a file? |
| 02:36.07 | Qwell | I think that was one of the problems with older firmware |
| 02:36.25 | _Soul_ | how to tell ? how to solve ? |
| 02:36.48 | [hC] | never dealt with that one yet |
| 02:36.50 | [hC] | how big is the file? |
| 02:36.56 | _Soul_ | here's the log from the second tftpd server: http://pastebin.com/488097 |
| 02:37.06 | _Soul_ | 476KB |
| 02:37.20 | *** part/#asterisk lrizzo (n=luigi@host114-164.pool8259.interbusiness.it) |
| 02:37.23 | Qwell | I'd try a 5.x firmware |
| 02:37.35 | Qwell | or a newer sccp maybe |
| 02:37.37 | _Soul_ | don't have the files, can anybody help ? |
| 02:37.46 | _Soul_ | alas, dont have ANY files ;) |
| 02:37.50 | [hC] | I have a P0S30203.bin thats 124kb |
| 02:38.03 | Qwell | 2.0.3 is OLD, heh |
| 02:38.05 | _Soul_ | [hC], can i try that ? |
| 02:38.13 | Qwell | it's probably older than what you've got now :p |
| 02:38.20 | _Soul_ | ugh, ok ;) |
| 02:38.30 | [hC] | im not sure but i think im using it on my phones |
| 02:38.38 | [hC] | and i dont think its 2.0.3 |
| 02:38.40 | Qwell | -rw-r--r-- 1 root root 128996 Dec 7 13:01 P00307020300.bin |
| 02:38.41 | [hC] | I think its just NAMED that. |
| 02:39.06 | _Soul_ | this is like the chicken and the egg, cant have the firmware cos dont have the firmware |
| 02:39.17 | Qwell | I bet that's why it says to go to a newer sccp first |
| 02:39.36 | [hC] | ah its 7.2 |
| 02:39.38 | [hC] | not 2.3 |
| 02:39.41 | [hC] | 03-07-02-00 |
| 02:39.43 | Qwell | neat |
| 02:39.47 | _Soul_ | hc, so should we try that ? |
| 02:40.13 | Qwell | _Soul_: is it the .bin or .sbn that's 475k? |
| 02:40.29 | Qwell | -rw-r--r-- 1 root root 592222 Jul 5 19:55 P0S3-07-4-00.bin |
| 02:40.33 | Qwell | nm, heh |
| 02:40.41 | _Soul_ | Qwell, both |
| 02:40.45 | Qwell | yeah |
| 02:42.01 | *** join/#asterisk Flauto (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 02:42.15 | Flauto | hey people |
| 02:42.18 | Flauto | happy new year |
| 02:42.25 | Flauto | i was installing a2billing |
| 02:42.25 | _Soul_ | i'm not really interested in the latest and greatest version, just a stable version that you guys use and approve |
| 02:42.53 | Flauto | but it does not seem that i can make it |
| 02:43.02 | Flauto | is there anyone can give me a hand |
| 02:45.27 | *** join/#asterisk Katty (n=angela@ppp-70-255-38-119.dsl.stlsmo.swbell.net) |
| 02:45.37 | Katty | hihi |
| 02:46.03 | cripito | hi katty |
| 02:46.38 | Katty | twisted: mew? |
| 02:46.53 | Katty | i could actually use support on something /asterisk/ related for a change. |
| 02:47.34 | Katty | how weird is that. |
| 02:49.25 | _DAW-LAPTOP | hello |
| 02:51.16 | Ariel_ | Katty, hello how are you tonight? |
| 02:51.35 | _Soul_ | hc, qwell ? |
| 02:52.00 | *** join/#asterisk annonimous (i=annonimo@201.135.196.52) |
| 02:52.05 | annonimous | hello! |
| 02:52.17 | Katty | Ariel_: fine thanks...asterisk is giving me heartburn though |
| 02:52.31 | Ariel_ | Katty, so what is the issue |
| 02:53.07 | Katty | Ariel_: well, basically, we can recieve calls all day long... |
| 02:53.19 | Katty | Ariel_: but when asterisk picks up, there's nothing but static. |
| 02:53.48 | Ariel_ | zap/t1 pri?? |
| 02:54.02 | Katty | Ariel_: analog lines |
| 02:54.27 | Ariel_ | ok are they connect to what card? |
| 02:54.28 | Katty | Ariel_: i wish i just knew where to start. |
| 02:54.43 | Katty | Ariel_: they're 4 port cards. |
| 02:54.52 | Katty | Ariel_: and we have 8 ports total, on two cards. |
| 02:55.14 | Ariel_ | ok they were working before correct |
| 02:55.20 | Ariel_ | can you make outbound calls? |
| 02:56.45 | Katty | i /think/ so, but i can't seem to pull up our company website or ssh to my box. |
| 02:57.08 | Ariel_ | hummm |
| 02:57.14 | Ariel_ | is there anyone there? |
| 02:57.20 | Katty | Ariel_: i don't think we can, now that i think about it |
| 02:57.27 | Katty | Ariel_: nah, it's closed down for the night. |
| 02:57.36 | Ariel_ | bummer |
| 02:57.47 | Katty | Ariel_: and all the complaints will come in at 7:30 >.< |
| 02:57.48 | Ariel_ | you did not do any updates or changes have you? |
| 02:58.12 | Katty | not that i can think of |
| 02:59.00 | Ariel_ | 7:30 am mountain time? |
| 02:59.45 | *** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net) |
| 03:00.27 | Katty | central ;) |
| 03:00.55 | Ariel_ | I would suggest you find a way to reboot it before then. |
| 03:01.22 | Katty | oh, i have. |
| 03:01.22 | Ariel_ | if you can attach a ananlog phone to the line to make sure it's not hthe actual line that is bad |
| 03:01.40 | Katty | and i made sure that it was asterisk...plugged a regular phone into the lines |
| 03:01.42 | Katty | works fine |
| 03:02.15 | Ariel_ | have you changed the lines around? could it be a bad module. I have seem them go bad |
| 03:02.28 | Qwell | _Soul_: ? |
| 03:02.42 | brockj49464 | Anybody got a working config for Zingotel? |
| 03:02.44 | Katty | Ariel_: a bad module? |
| 03:02.47 | *** join/#asterisk locid (n=locid@206-248-133-11.dsl.teksavvy.com) |
| 03:02.57 | Katty | Ariel_: like a messed up driver, you mean? |
| 03:03.09 | Ariel_ | Katty, the tdm400 board has 4 little modules |
| 03:03.16 | Ariel_ | sometimes one can go bad |
| 03:03.28 | locid | is agents.conf needed in realtime config? |
| 03:03.36 | Qwell | locid: only if you use agents |
| 03:03.56 | annonimous | hello, sorry, anybody here where can i found a good how to of asterisk? (i need to setit up with a spa 3000 (cause i dont have money for the trunk pci card =() |
| 03:04.09 | Qwell | ~wikis |
| 03:04.10 | jbot | somebody said wikis was http://www.voip-info.org |
| 03:04.12 | Qwell | annonimous: ^ |
| 03:04.20 | Katty | Ariel_: let me call in on another line |
| 03:04.24 | Ariel_ | ~docs |
| 03:04.26 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:04.26 | locid | for call queueing, i need agents, therefore agents.conf is needed? |
| 03:04.36 | Qwell | locid: if you need agents, yes |
| 03:04.37 | annonimous | wikis? |
| 03:04.58 | locid | qwell: can you have call queuing without agents? |
| 03:05.01 | Ariel_ | annonimous, the wiki is a good start also the asteriskdocs.org |
| 03:05.04 | Qwell | locid: sure |
| 03:05.09 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 03:05.37 | annonimous | Ariel_ i see, thank you =D |
| 03:05.37 | locid | who answers the calls? |
| 03:05.41 | Qwell | locid: queue members |
| 03:06.10 | locid | difference between queue members and agents? |
| 03:06.19 | Katty | Ariel_: same static.. |
| 03:06.21 | Qwell | agents suck. queue members suck also |
| 03:06.25 | Qwell | queue members suck less |
| 03:06.33 | Qwell | dynamic queue members suck the least |
| 03:06.37 | Qwell | (but have the most bugs...) |
| 03:06.44 | Katty | Ariel_: if i give you our 800 number, will you give it a call it and listen to this static thing? |
| 03:06.50 | Ariel_ | Katty, is that line on the same board? |
| 03:07.01 | annonimous | thanks everybody ill read it =D! |
| 03:07.06 | Katty | Ariel_: hrmm |
| 03:07.11 | Katty | Ariel_: now that i don't know |
| 03:08.13 | *** join/#asterisk SwK_ (n=SwK@12-219-151-128.client.mchsi.com) |
| 03:08.18 | Ariel_ | Katty, when are you going to be there? |
| 03:09.06 | Katty | Ariel_: tomorrow, around 8 central (= |
| 03:09.31 | Ariel_ | ahh 30 minutes after the yelling starts |
| 03:09.35 | Katty | nah |
| 03:09.39 | Katty | i'll be there at 7:30 |
| 03:09.44 | Katty | people start getting in around 8 |
| 03:09.58 | Katty | Ariel_: let's pretend my two numbers are on different boards. |
| 03:10.12 | Katty | Ariel_: and there's horrid staticy stuff on both boards. |
| 03:10.35 | Ariel_ | check irq's has the server rebooted? |
| 03:10.56 | robl^ | PCI 2.2 compliant? |
| 03:10.57 | Katty | i rebooted the server earlier today. |
| 03:11.09 | Katty | but i'lll check what irqs they're on ariel |
| 03:12.09 | robl^ | I've seen that happen also if the power supply was a little flakey and had some power fluctuations |
| 03:12.27 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 03:12.28 | Katty | k, robl^ (= |
| 03:12.32 | Katty | i'll take a look at that too |
| 03:14.32 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 03:17.12 | [hC] | k gonna take off for a bit and go get dinner |
| 03:17.19 | [hC] | chow kids |
| 03:17.32 | *** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com) |
| 03:19.19 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:20.40 | *** join/#asterisk blitzrage (n=blitzrag@12.174.37.165) |
| 03:22.02 | [TK]D-Fender | exit |
| 03:22.37 | Thazza | Hey all.. I have a question. |
| 03:22.52 | trixter | http://trekweb.com/articles/2006/01/02/43b963f9d031c.shtml VoIPPowered Tech Brings ST:TNG Combadge Treknology to Workplaces -- dear god there is no hope |
| 03:24.36 | Thazza | Everytime i try and connect to my running asterisk server, i get the following message. |
| 03:24.49 | Thazza | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 03:24.59 | Qwell | Thazza: does /var/run/asterisk.ctl exist? |
| 03:25.08 | Thazza | Yet if i do a ps -A asterisk is in the menu. |
| 03:25.18 | Qwell | Thazza: does /var/run/asterisk.ctl exist? |
| 03:25.21 | SpaceBass | anyone know of a provider (sip or iax) that allows setcalleridname ? |
| 03:25.25 | Qwell | SpaceBass: none |
| 03:25.33 | Qwell | SpaceBass: cidname is looked up at the remote end |
| 03:25.38 | Thazza | Qwell: Well yes and no.. there is a file called /var/run/asterisk.ctl yet the size is 0 bytes. |
| 03:25.40 | *** join/#asterisk Flauto (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
| 03:25.44 | SpaceBass | Qwell: that explains a lot... thanks! |
| 03:25.45 | Qwell | Thazza: that's fine |
| 03:25.58 | Qwell | Thazza: what user are you trying to do this as? |
| 03:26.06 | Qwell | They need permission to that file (and others, no doubt) |
| 03:26.40 | Thazza | Qwell: well i am logged in as root, and trying to run asterisk -r and the /var/run/asterisk.ctl is owned and group by root. |
| 03:27.01 | Lloydie-t | I going to re-install *. someone mentioned a directory to delete before I do it |
| 03:27.08 | Lloydie-t | is it /usr/lib/asterisk/modules? |
| 03:27.13 | *** join/#asterisk lrizzo (n=luigi@host114-164.pool8259.interbusiness.it) |
| 03:27.15 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 03:27.29 | *** join/#asterisk emrah (n=emrah@knsrv1-zrh8048.net1.kavun.ch) |
| 03:27.35 | xbmodder_lappy | hey |
| 03:27.39 | syle | happy new year! |
| 03:27.45 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 03:27.48 | xbmodder_lappy | anyone here have experience with asterisk and Sipura devices |
| 03:27.49 | xbmodder_lappy | ? |
| 03:28.12 | syle | lloydie-t: yes , stupid that asterisk make install don;t replace with new ones to start with :) |
| 03:28.46 | Qwell | syle: it does replace the new ones. What it doesn't do, is delete old ones |
| 03:28.46 | Qwell | It can't possibly know which are third party, and which are it's own |
| 03:28.55 | syle | you mean replace the old ones? |
| 03:29.00 | Qwell | no |
| 03:29.01 | *** part/#asterisk lrizzo (n=luigi@host114-164.pool8259.interbusiness.it) |
| 03:29.02 | Lloydie-t | Thanks chaps |
| 03:29.04 | Qwell | I mean delete the old ones |
| 03:29.22 | *** join/#asterisk annonimous (i=annonimo@201.135.196.52) |
| 03:29.39 | Thazza | Qwell: any ideas on why it will not let me run? |
| 03:29.39 | Qwell | for instance chan_modem.so |
| 03:29.49 | Qwell | Thazza: nope... |
| 03:30.14 | Qwell | Thazza: You're sure asterisk is running properly? |
| 03:30.52 | annonimous | hello all |
| 03:31.07 | syle | i think your wrong |
| 03:31.14 | syle | check Makefile line 702 |
| 03:31.22 | syle | oldmodcheck: |
| 03:31.27 | Qwell | okay, chan_modem was a bad choice |
| 03:32.06 | *** join/#asterisk dd (n=dd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 03:32.09 | Qwell | no |
| 03:32.15 | Qwell | That doesn't delete them. it only displays them |
| 03:32.31 | Qwell | like I said, it can't know which ones are thirdparty, and which are it's own |
| 03:32.47 | Qwell | it would be very bad to delete all of those files |
| 03:33.07 | Qwell | however, there is stuff in the works to deal with that |
| 03:33.48 | syle | holdon checking source code |
| 03:34.06 | Qwell | I know for certain I'm right on this one. :) |
| 03:34.17 | syle | ok yea it does replace old ones, but leaves third party ones alone warning you about them to be exact |
| 03:34.28 | Qwell | yes, that's what I said. :p |
| 03:35.53 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:36.27 | syle | ok i just didn;t understand syle: it does replace the new ones. <--didn;t make any sense to me , you either replace old ones or don;t makes proper english sense to me :) |
| 03:36.43 | Lloydie-t | should I compile asterisk-addons before asterisk? |
| 03:37.09 | syle | lloydie no |
| 03:37.23 | syle | however you should do zaptel and libpri first |
| 03:37.34 | Qwell | libpri before zaptel before asterisk, everything else after |
| 03:37.38 | syle | sorry libpri then zaptel then asterisk then asterisk-addons in that order |
| 03:37.44 | syle | causes less problems |
| 03:37.48 | Qwell | indeed |
| 03:38.40 | Lloydie-t | Oh i've done zaptel and then libpri, as per digium instructions on web |
| 03:39.21 | syle | if you want to know why, well libpri compiles some header files and zaptel attempts to use them, and if zaptel looking for the new libpri routines and if it don;t find them, compile error basically |
| 03:39.55 | Qwell | Lloydie-t: can you show me where on the instructions is says to do zaptel before libpri? |
| 03:40.03 | Qwell | If that is the case, I'll see about getting it corrected... |
| 03:40.28 | Thazza | Qwell: Sorry was looking at config.. Yet it is a live system running. answering calls, and making them fine. |
| 03:41.06 | Qwell | Thazza: is asterisk running as root? |
| 03:41.14 | Lloydie-t | I mean asterisk.org 'http://www.asterisk.org/download' |
| 03:41.51 | Qwell | hmm, so it does |
| 03:41.53 | syle | hey qwell: ever heard of this, some other people on analog call me and cannot use dtmf from their systems to mine, this has happened to me with some doctors and dentists offices with their own pbx;s |
| 03:42.08 | Qwell | syle: everything is analog? |
| 03:42.11 | syle | normal analog callers don;t have a problem |
| 03:42.15 | syle | yep |
| 03:42.35 | syle | i use a rhino channel bank at home-> T1 card-> to my asterisk box at home |
| 03:42.49 | Qwell | I think I have...lemme think |
| 03:43.04 | *** join/#asterisk ManxPowe (n=ewieling@24-179-48-91.static.slid.la.charter.com) |
| 03:43.18 | Qwell | okay, yeah... in zapata.conf, try to set relaxdtmf=yes |
| 03:44.10 | *** join/#asterisk emike240s (n=mike240s@ool-44c45c3f.dyn.optonline.net) |
| 03:45.11 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
| 03:45.13 | syle | alright tried it thanks |
| 03:45.38 | syle | be nice if hospital/doctors etc could get through to me :) |
| 03:47.51 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 03:48.17 | Lloydie-t | has anyone compiled asterisk-addons/res_sqlite3? I got a world of errors before I started this re-install |
| 03:50.46 | *** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 03:51.07 | syle | pastebin is your friend |
| 03:51.11 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:51.34 | *** join/#asterisk bmg505 (n=leon@dsl-146-23-215.telkomadsl.co.za) |
| 03:52.04 | syle | hey qwell: also have another problem with calling walmart , my normal sequence is 3-1-1 to get electronics dept. , when it starts ringing finally and someone picks up, line disconnects on me |
| 03:52.08 | syle | ever heard of that? |
| 03:52.28 | syle | works usually after 5th try |
| 03:52.47 | syle | pisses me off cause their is no debugging on this, maybe i should call rhino lol |
| 03:53.10 | Lloydie-t | syle, I'll try again once I have finished the install and paste if I get any problems |
| 03:53.21 | *** join/#asterisk iiii (i=licucude@ool-44c784a0.dyn.optonline.net) |
| 03:53.38 | Qwell | syle: using callprogress? |
| 03:53.47 | syle | i checked that, nope |
| 03:53.50 | Qwell | I hear it sucks |
| 03:53.51 | Qwell | k |
| 03:54.01 | Qwell | busydetect or anything? |
| 03:54.32 | syle | its commented out, i think it may be defaulting to yes, unsure |
| 03:54.37 | Qwell | no clue |
| 03:55.11 | syle | oww wait it is on |
| 03:55.15 | syle | on wrong asterisk box |
| 03:55.20 | Qwell | heh |
| 03:55.29 | Qwell | callprogress? |
| 03:55.35 | syle | that is off |
| 03:55.37 | Qwell | oh |
| 03:55.52 | syle | well i;ll turn busydetect off see how that goes |
| 03:55.57 | Qwell | I forget the exact reason, but I hear busydetect can cause problems |
| 03:56.36 | *** join/#asterisk linagee (n=linagee@netblock-68-183-1-149.dslextreme.com) |
| 04:00.02 | syle | hmmm well i got through that time |
| 04:00.13 | syle | maybe a fluke or maybe it worked |
| 04:00.48 | syle | i remember why i turned it on now, i was programming something before to detect answering machines and i need that on |
| 04:00.55 | syle | but thanks |
| 04:01.46 | syle | i remember from my tests on that project though callprogress would actually cause random hangups on me for no reason....so i hear you on that one :) |
| 04:03.13 | Lloydie-t | bloody hell. first res_sqlite3 compile error 'make: execvp: /usr/src/asterisk/contrib/scripts/astxs: Permission denied' |
| 04:03.52 | syle | are you root? |
| 04:04.31 | Lloydie-t | I logged in as root |
| 04:05.12 | syle | i never heard of anyone using sqlite3 over mysql or postgres |
| 04:05.56 | syle | i think you need to paste more than that on pastebin |
| 04:06.05 | Lloydie-t | OK |
| 04:06.15 | syle | if your root you should never have gotten permission denied so need more debugging info |
| 04:06.51 | X-Rob | chmod 755 /usr/src/asterisk/contrib/scripts/astxs |
| 04:07.03 | X-Rob | ls -l /usr/src/asterisk/contrib/scripts/astxs even |
| 04:07.04 | *** join/#asterisk jasonwolfe0u812 (n=jasonwol@adsl-072-151-106-082.sip.asm.bellsouth.net) |
| 04:07.05 | syle | yeah thats a good point |
| 04:07.11 | syle | it might not be executable |
| 04:07.26 | Lloydie-t | syle http://pastebin.ca/35579 |
| 04:08.07 | syle | yep xrob is right from your pastebin |
| 04:08.40 | *** join/#asterisk Strom_C_ (i=strom@66.159.243.60) |
| 04:09.06 | syle | if that don;t work paste the first line from astxs |
| 04:09.51 | syle | unless its a c program :) |
| 04:10.56 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 04:12.44 | *** join/#asterisk lll (i=joelsola@202.160.161.93) |
| 04:15.25 | Lloydie-t | Hmmm, I might have got there. not sure?? have a look chaps http://pastebin.ca/35581 |
| 04:18.16 | syle | says it installed fine |
| 04:18.25 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 04:18.35 | Qwell | mogmogmogmogmogmog! |
| 04:18.43 | mogorman | qwell! |
| 04:18.50 | Qwell | aww, only one? I see how it is |
| 04:18.54 | mogorman | so im not happy |
| 04:18.57 | syle | so anything new happen with asterisk in last 2 weeks i was away lol |
| 04:18.57 | mogorman | i have work tommorrow |
| 04:19.04 | Qwell | shouldn't you be happy? |
| 04:19.06 | syle | yeah welcome back to work |
| 04:19.07 | syle | :( |
| 04:19.09 | syle | hehee |
| 04:19.10 | mogorman | syle like you wouldnt believe |
| 04:19.12 | Qwell | are those 5 days finally up? |
| 04:19.14 | mogorman | yeah i am |
| 04:19.20 | mogorman | but the whole getting dressed thin |
| 04:19.20 | mogorman | g |
| 04:19.23 | mogorman | major downer |
| 04:19.25 | Qwell | wtf |
| 04:19.27 | syle | or out of bed |
| 04:19.29 | Qwell | Mark makes you dress? :) |
| 04:19.29 | syle | hahaaa |
| 04:19.31 | mogorman | i have been in pajamas since thursday |
| 04:19.35 | mogorman | well in clothes |
| 04:19.37 | Qwell | tsk, tsk, tsk...heh |
| 04:19.39 | mogorman | like tshirt pants |
| 04:19.47 | mogorman | but not real "work dressed" |
| 04:19.59 | Qwell | unless you've got "visitors"? |
| 04:20.04 | syle | yeah jeans are so uncomfortable hehe |
| 04:20.07 | mogorman | not even then |
| 04:20.10 | Qwell | cool |
| 04:20.12 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 04:20.21 | mogorman | at least for people in my sector |
| 04:20.21 | Qwell | I used to hate dressing up for work... |
| 04:20.30 | mogorman | yeah its annoying |
| 04:20.33 | Qwell | I had to do that for about a year |
| 04:20.41 | Katty | i rather like dressing up for work |
| 04:20.44 | Qwell | Then I just gradually stopped |
| 04:20.47 | syle | i did it for a year and a half! |
| 04:20.52 | Qwell | Katty: I'm sure you love it |
| 04:20.52 | mogorman | well i have nothing against it |
| 04:20.57 | mogorman | i just hate changing my routine |
| 04:21.09 | Qwell | most women do... (yes, I'm generalizing...sorry) |
| 04:21.24 | Katty | i do on special occasion, but in all reality, blue jeans are not soft. |
| 04:21.28 | syle | woman should have to, we have to have something to drool over |
| 04:21.30 | syle | :) |
| 04:21.49 | NewSole | or on |
| 04:21.50 | Katty | we're not here for the sole purpose of entertaining males. |
| 04:22.05 | syle | really? damn i got to tell my wife that she will be pleased |
| 04:22.07 | implicit | ok |
| 04:22.14 | Katty | you get any male around me with that attitude and i'm ready to claw things. |
| 04:22.16 | Qwell | syle: DON'T SPREAD IT AROUND! |
| 04:22.21 | Qwell | wtf are you thinking? :p |
| 04:22.22 | syle | lol |
| 04:22.44 | Qwell | okay, I'm done joking around with Katty today |
| 04:22.46 | mogorman | heh |
| 04:22.48 | mogorman | sorry katty |
| 04:23.03 | NewSole | oowwww claws...... cat scrach fever |
| 04:23.03 | implicit | Katty, and how about a female with that attitude? |
| 04:23.08 | implicit | Katty, sexist bitch |
| 04:23.16 | syle | yeah your kind of outnumbered no offence intended :) |
| 04:23.16 | implicit | Katty, better yet, cunt |
| 04:23.38 | implicit | (calling a girl cunt always makes them crazy) |
| 04:23.38 | rob0 | and implicit means that in the nicest possible way, I'm sure :) |
| 04:23.46 | implicit | rob0, for sure |
| 04:24.03 | syle | i called my wife a cunt once, and i never heard the end of it, very bad thing to say to a woman :) |
| 04:24.04 | idpromnut | <PROTECTED> |
| 04:24.08 | mogorman | lucky i dont have ops in this channel |
| 04:24.14 | mogorman | people would be getting kicked... |
| 04:24.21 | implicit | i know! |
| 04:24.33 | mogorman | heh like 1/6th of the people in asterisk-dev have ops |
| 04:24.48 | rob0 | Hi Katty, how are you? |
| 04:24.51 | syle | i asked why this is, this is cause mom;s teach their daughters that is the worst possible word |
| 04:24.53 | Qwell | 1 is chanserv, so more like 1/7 :p |
| 04:24.54 | implicit | mogorman, makes kicking quicker |
| 04:25.07 | mogorman | heh |
| 04:25.13 | mogorman | i am quick to kick |
| 04:25.23 | ManxPowe | My opinion is that, like many people on this channel, Katty can take offense a bit too easily, but she's pretty cool. |
| 04:25.24 | implicit | i know |
| 04:25.42 | mogorman | yeah true |
| 04:25.43 | implicit | ManxPowe, yeah |
| 04:25.50 | syle | at least i admit i am wrong when i am, alot of people in here don't :) |
| 04:25.53 | mogorman | i also just enjoy kicking |
| 04:26.54 | syle | so i guess nothing new in last couple weeks, everyone was to busy drinking to lol |
| 04:26.55 | rob0 | I think Katty is offended this time |
| 04:27.13 | rob0 | or plotting revenge |
| 04:27.46 | mogorman | heh |
| 04:28.13 | syle | naw i think she spends 20 min reading emails :) |
| 04:28.31 | idpromnut | anybody have a link for a good PRI primer; preferrably wrt Asterisk, but I'll take what I can get :) |
| 04:28.39 | rob0 | I just hope I'm here when she comes back |
| 04:29.01 | Qwell | ~wikis |
| 04:29.02 | jbot | it has been said that wikis is http://www.voip-info.org |
| 04:29.05 | Qwell | idpromnut: should be something there |
| 04:29.08 | mogorman | cisco docs idpromnut if you want real pri info |
| 04:29.10 | rob0 | this might be better then New Year's fireworks |
| 04:29.15 | syle | anyone get a xbox360 over holidays? |
| 04:29.17 | jasonwolfe0u812 | can someone recomend a good softphone for use with IAX |
| 04:29.18 | mogorman | asterisk setup is like 5 min long |
| 04:29.23 | idpromnut | Qwell: been through there; but I'll have another go :) |
| 04:29.25 | Qwell | syle: yeah, I'm running asterisk on mine already |
| 04:29.26 | mogorman | what os jasonwolfe0u812 |
| 04:29.30 | rob0 | jasonwolfe0u812: KIAX |
| 04:29.32 | jasonwolfe0u812 | windows |
| 04:29.33 | syle | lol |
| 04:29.46 | ManxPowe | All Softphones suck! |
| 04:29.50 | rob0 | true |
| 04:29.53 | syle | i been following the hacking project, they are comming along quite well, linux will be on it soon enough |
| 04:29.57 | mogorman | iaxphone jasonwolfe0u812 |
| 04:30.06 | mogorman | really syle |
| 04:30.07 | Qwell | syle: I sure as hell wouldn't want to compile anything on it |
| 04:30.12 | Qwell | or...even...you know...play games |
| 04:30.12 | mogorman | i havent heard anything of promise |
| 04:30.41 | Qwell | I wonder... |
| 04:30.43 | syle | yeah they already got patches into the 2.4.x kernels for the filesystem on it |
| 04:30.52 | Qwell | Do you think I could get away with naming a softphone "Micro"? |
| 04:30.58 | Qwell | people would call it the Micro softphone |
| 04:31.07 | syle | fatfx or something |
| 04:31.17 | mogorman | lol Qwell |
| 04:31.19 | mogorman | you should do it |
| 04:31.57 | Qwell | uh oh |
| 04:32.12 | mogorman | lol |
| 04:32.22 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-70-93-239-175.san.res.rr.com) |
| 04:32.34 | Qwell | I read a bash.org post that started just like this |
| 04:32.44 | mogorman | ? |
| 04:32.49 | mogorman | i love bash |
| 04:32.50 | syle | well my biggest interest in xbox360 is getting rid of computer in my living room and making that a media center , somehow nfs mounting or samba all my movies from it |
| 04:32.58 | mogorman | every now and then i benge on it |
| 04:33.00 | Qwell | might've been qdb |
| 04:33.24 | Qwell | http://qdb.us/48894 |
| 04:33.38 | Qwell | somehow, her leaving reminded me of that :p |
| 04:34.04 | mogorman | heh that has happened with me and my woman |
| 04:34.07 | Qwell | heh |
| 04:34.13 | mogorman | but she is only 2 min away |
| 04:34.21 | mogorman | so it was more of a shock when she nocked on door |
| 04:34.23 | *** join/#asterisk Administrator (n=Administ@202.57.0.45) |
| 04:34.48 | syle | oww god , never live next to your girlfriend |
| 04:34.53 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 04:34.55 | syle | thats asking for trouble :) |
| 04:35.11 | idpromnut | Qwell: yup, that would be my lady to the letter :) |
| 04:35.14 | mogorman | i love my woman |
| 04:35.25 | Dandan | lol |
| 04:35.32 | Dandan | i even sleep next to one :) |
| 04:35.33 | Administrator | helo |
| 04:35.33 | *** join/#asterisk PBXtech (i=nik@248.sub-70-213-205.myvzw.com) |
| 04:35.37 | syle | what happens if you bring another one home and she comes over unexpected |
| 04:35.40 | Dandan | not talking about living :) |
| 04:35.56 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
| 04:36.01 | Administrator | i'm newbie |
| 04:36.01 | riddlebox | is there a way to test that when a voicemail is left, that you can see asterisk try to send the email out to the user? |
| 04:36.07 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 04:36.12 | mogorman | why would i cheat on my woman? |
| 04:36.16 | Qwell | riddlebox: You want to see it try? |
| 04:36.25 | Qwell | mogorman: Why would you, or why would others? |
| 04:36.32 | riddlebox | Qwell:for some reason it is not sending |
| 04:36.32 | Qwell | (I don't have an answer for either) |
| 04:36.35 | mogorman | lol |
| 04:36.38 | idpromnut | Administrator: you know you should never log onto IRC as Administrator; people get their boxes hacked that way |
| 04:36.43 | syle | mogorman you have to be married at least 2 years with one before you can comment on a woman :) |
| 04:37.15 | Administrator | oh ya h to change |
| 04:37.16 | mogorman | we are happily crazy as hell |
| 04:37.23 | syle | thats cause you don;t live together :) |
| 04:37.32 | Nugget | heh |
| 04:37.44 | litecode | is there a stable/scalable/usable multiple parking lot system available someplace? |
| 04:37.48 | litecode | for 1.2? |
| 04:37.55 | syle | no shit |
| 04:37.58 | syle | wow |
| 04:38.12 | syle | is she ever home? |
| 04:38.13 | syle | hehe |
| 04:38.13 | mnemonic | oh okay |
| 04:38.25 | mogorman | heh |
| 04:38.27 | Qwell | litecode: I think oej has a branch for something like that |
| 04:38.29 | idpromnut | syle: she's sleeping in the next rrom as we speak |
| 04:38.41 | idpromnut | s/rrom/room/ |
| 04:38.45 | syle | kids? |
| 04:38.52 | idpromnut | nah, you jest :) |
| 04:39.28 | Mnemonic | helo all i'm newbie |
| 04:39.36 | implicit | i know |
| 04:39.37 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 04:39.50 | brockj49464 | Anyone have Zingotel working with Asterisk? |
| 04:39.59 | Mnemonic | ;-) |
| 04:40.24 | syle | 6 years and no kids |
| 04:40.27 | syle | something wrong there |
| 04:40.34 | syle | impotent? |
| 04:40.37 | Mnemonic | bye bye s U |
| 04:41.54 | mogorman | lol |
| 04:42.03 | Nugget | what's wrong with not having kids? |
| 04:42.08 | Nugget | I'm never going to have any |
| 04:42.19 | *** join/#asterisk Administrator (n=Administ@202.57.0.45) |
| 04:42.21 | ManxPowe | I'm never having kids either |
| 04:42.28 | idpromnut | syle: I'm 26. Gimme a bit of time ;) |
| 04:42.43 | syle | yeah i use to say that, but i go thoroughbred |
| 04:42.57 | Nugget | for me it's quite beyond "saying that". |
| 04:43.03 | syle | no raincoat for this stallion |
| 04:43.05 | Nugget | I've already taken the necessary precautions to prevent it. |
| 04:43.46 | syle | i use to to |
| 04:43.52 | syle | my name is john doe |
| 04:43.53 | syle | haha |
| 04:44.18 | Qwell | Nugget: precautions? |
| 04:44.22 | Qwell | Nugget: Can I guess? |
| 04:44.27 | Qwell | You...hmm |
| 04:44.29 | Qwell | use asterisk? :) |
| 04:44.39 | syle | hahaa |
| 04:44.46 | Nugget | hah |
| 04:44.48 | syle | are you saying he has no life! |
| 04:44.56 | Qwell | syle: I'm saying none of us do. |
| 04:44.57 | Qwell | :P |
| 04:44.58 | ManxPowe | none of us have lives. |
| 04:45.00 | Nugget | nah, I got snipped a few years ago. :) |
| 04:45.06 | Qwell | Nugget: eeps |
| 04:45.10 | ManxPowe | Nugget, good for you. |
| 04:45.12 | mogorman | i always liked the whats the best contraception |
| 04:45.16 | mogorman | a woman laughing at you |
| 04:45.28 | *** join/#asterisk Blankman (n=kvirc@c-24-61-183-130.hsd1.nh.comcast.net) |
| 04:45.31 | syle | i use to think i did, but one dayi mentioned the mythtv box in my livingroom and got classified as a nerd hehe |
| 04:45.34 | ManxPowe | Doesn't help with STDs, but it's a good thing to do if you don't want kids. |
| 04:45.37 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:45.52 | Blankman | Hey guys. |
| 04:45.58 | Qwell | Blankman: morning |
| 04:45.59 | Nugget | yeah, it's quite a relief to have finally done it. |
| 04:46.20 | Lloydie-t | I did the res_sqlite3 wrong first time round. I should have done make before make install. got this problem |
| 04:46.20 | Lloydie-t | In file included from res_sqlite.c:14: |
| 04:46.20 | Lloydie-t | In file included from res_sqlite.c:28: |
| 04:46.20 | Lloydie-t | sqlite-3.2.8/sqlite3.h:1254: warning: function declaration isn't a prototype |
| 04:46.21 | Lloydie-t | make: *** [res_sqlite.so] Error 255 |
| 04:46.24 | Blankman | Hey Qwell ... not morning in est land yet ;-) |
| 04:46.33 | Qwell | Blankman: yeah, I never do get it right |
| 04:46.47 | syle | asterisk omg, if i knew how much was involved before starting this, i don;t know if i would have done it, you have a 6 months of reading and playing to do to get out of newbie state |
| 04:47.13 | Lloydie-t | I meant http://pastebin.ca/35586 |
| 04:47.50 | Blankman | Hey, I have a question. I am traveling all over targh ... the new kvirc doesn't do tab complete! |
| 04:47.56 | syle | well 3 if you can program |
| 04:47.57 | Blankman | I will be back :-) |
| 04:47.58 | Blankman | exit |
| 04:48.03 | Blankman | opps :-) |
| 04:49.04 | riddlebox | Qwell:I guess I will have to do more reading |
| 04:49.06 | ManxPowe | He needs to switch to xChat |
| 04:49.15 | syle | xchat wtf |
| 04:49.18 | *** join/#asterisk Blankman (n=blankman@c-24-61-183-130.hsd1.nh.comcast.net) |
| 04:49.35 | Blankman | Qwell: k. lets try that again :-) |
| 04:49.43 | syle | you running linux as a client? |
| 04:50.17 | Lloydie-t | syle 6 months! I might as well give up. I don't even know linux yet. |
| 04:50.22 | syle | let me save you the heartache, windowsXP + securecrt + flashfxp , and you can still play games! |
| 04:51.04 | syle | owww man |
| 04:51.14 | syle | i would learn linux first |
| 04:51.32 | syle | most people in here already been using linux for like 10 years |
| 04:52.35 | syle | more stuff you install the more you learn it, so this is good exp for you to i guess |
| 04:53.36 | Nugget | and eventually you'll learn that linux sucks and move on to freebsd. :) |
| 04:53.43 | Nugget | but give it time. |
| 04:53.57 | rob0 | Nugget, I haven't learned that yet :) |
| 04:54.00 | syle | till you realize the zaptel drivers suck for it and time is all it will ever be |
| 04:54.01 | Lloydie-t | gotta start somewhere and its is good for me to have a project to help me learn |
| 04:54.11 | Qwell | Then you'll realize that FreeBSD is overrated, and move back to BeOS |
| 04:54.41 | syle | i don;t know BeOS can compete with freebsd's ports collection :) |
| 04:54.50 | Nugget | Qwell: http://gallery.distributed.net/sanfran3/be_everyone |
| 04:55.20 | Qwell | Nugget: okay, I take back everything I said about BeOS |
| 04:55.28 | Nugget | heh |
| 04:55.41 | Qwell | bunch of ugly fellows :p |
| 04:55.48 | Nugget | can't argue with that ;) |
| 04:55.52 | Qwell | Especially that one on the left. ;) |
| 04:56.01 | mogorman | man beos was awesome |
| 04:56.03 | syle | i;ve never heard of beos |
| 04:56.14 | idpromnut | beos++!!! |
| 04:56.17 | syle | what kind of operating system is that |
| 04:56.21 | mogorman | i am kinda interested in what haiku people come up with |
| 04:56.24 | Lloydie-t | I got the red hat linux bible. What a pile of poo. not enough command line info |
| 04:56.26 | mogorman | it was a "media os" |
| 04:56.27 | idpromnut | wow, that goes back a couple of years. |
| 04:56.30 | Qwell | I think that guy on the right is a little out of place... |
| 04:56.33 | mogorman | its posix compliant |
| 04:56.46 | syle | posix, well should run asterisk then :) |
| 04:56.47 | mogorman | but it was really fast |
| 04:56.47 | Blankman | anyone have asterisk running on a laptop AND using iaxcomm to connect to it? |
| 04:56.53 | mogorman | probably could |
| 04:56.56 | Nugget | the guy on the far right is the only one in the photo who was actually a be, inc. employee |
| 04:56.59 | mogorman | but shouldnt |
| 04:57.02 | mogorman | its not a server os |
| 04:57.04 | Qwell | Nugget: odd |
| 04:57.05 | mogorman | its for workstations |
| 04:57.10 | mogorman | its all about being really fast |
| 04:57.11 | syle | yuk |
| 04:57.13 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 04:57.27 | syle | so the tcp stack is probably all misconfigured to hell to be a server default install |
| 04:57.49 | mogorman | so i couldnt tell you |
| 04:57.55 | Blankman | mogorman: it wasn't that beos was fast ... it was that beos file system rocked! |
| 04:58.12 | Blankman | mogorman: nothing like having a database for a filesystem :-) |
| 04:58.14 | syle | new filesystem kewl |
| 04:58.16 | mogorman | nothing was a bigger draw for me to beos than the responsiveness |
| 04:58.19 | syle | always like trying new ones |
| 04:58.23 | mogorman | it just was blazingly fast |
| 04:58.24 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:58.25 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:58.39 | mogorman | well it wasnt so much fast as when you clicked things they happened |
| 04:58.40 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:58.44 | mogorman | no wait time |
| 04:58.47 | Lloydie-t | any ideas on http://pastebin.ca/35586 ? |
| 04:58.49 | syle | i like reiserfs for databases, XFS for speed, JFS for high speed deletes |
| 04:58.52 | mogorman | very creative |
| 04:58.56 | Blankman | mogorman: also there "widget" factory was better than the sup'd up Next factories :-) |
| 04:59.07 | idpromnut | syle: check out HPFS+; metadata in the filesystem, and Apple is actually letting people populate it with their own metadata! :) |
| 04:59.10 | Qwell | Nugget: Wanna see a very scary looking Linux user? |
| 04:59.12 | mogorman | well i like gtk widgets |
| 04:59.20 | mogorman | i dont think anyone has ever done it better |
| 04:59.21 | Qwell | I'd run if I saw this guy in a dark alley |
| 04:59.39 | mogorman | but id love to see things get back to beos responsiveness |
| 04:59.46 | syle | apple fs's reminds me of hacking tivo's |
| 04:59.49 | Blankman | So no one dev'n on a laptop and a soft client? |
| 04:59.52 | syle | should try it |
| 05:00.00 | Nugget | Qwell: present company excepted? :) |
| 05:00.06 | Qwell | indeed |
| 05:00.13 | Qwell | see msg |
| 05:00.24 | mogorman | nah syle its oldddd |
| 05:00.28 | mogorman | wait for haiku |
| 05:00.48 | mogorman | i think i will start running it with haiku and linux |
| 05:00.53 | mogorman | on my new laptop |
| 05:01.03 | Qwell | haiku? |
| 05:01.11 | mogorman | haiku-os |
| 05:01.25 | mogorman | group of people who are doing an oss beos |
| 05:01.29 | Qwell | ahh |
| 05:02.02 | _Soul_ | Qwell, pvt ? |
| 05:02.10 | Qwell | _Soul_: sure |
| 05:02.38 | mogorman | its pretty ambituos project |
| 05:02.48 | mogorman | i imagine they will finish this year or the next |
| 05:02.59 | mogorman | or not finish |
| 05:03.08 | _Soul_ | Qwell, did you see my pvt message ? |
| 05:03.08 | mogorman | but have completely working system |
| 05:03.16 | Qwell | nope.. |
| 05:03.37 | syle | i think google is building their own fs as well, they have stated everyone could nfs mount their email space |
| 05:04.31 | mogorman | when did they let nfs? |
| 05:04.45 | syle | for awhile now, i think i mounted it once or twice |
| 05:04.55 | mogorman | googlefs != nfs |
| 05:04.56 | X-Rob | nfs? *boggle* |
| 05:05.02 | X-Rob | how incredibly insecure |
| 05:05.16 | _Soul_ | Qwell, i'm not sure whats wrong with my pvt queries with you, can you try to send me a pvt message ? |
| 05:05.21 | morale | its not nfs, he probably means a network file system of some sort. |
| 05:05.28 | mogorman | and letting people hack that out |
| 05:05.39 | syle | i mean network file system, custom client to mount it like nfs |
| 05:05.40 | mogorman | and supporting it are two very different things |
| 05:06.45 | syle | lots of google jobs lately |
| 05:06.50 | syle | i;lve noticed that |
| 05:06.58 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 05:07.17 | syle | i was thinking of getting one so i could get their source code to search rating algorithm hehe |
| 05:07.20 | syle | lol |
| 05:08.06 | mogorman | syle?!?! you = crazy? |
| 05:08.24 | Fubster | i think i might know what my audio problem is... |
| 05:08.28 | syle | if i had their source code to that i;d be a millionare within 6 months |
| 05:08.52 | mogorman | hmm sure |
| 05:08.54 | Fubster | my ztdummy is "UNCONFIGURED", according to the alarms |
| 05:09.24 | *** join/#asterisk venix (n=venix@CPE000625f5bd17-CM0011aea5246c.cpe.net.cable.rogers.com) |
| 05:09.26 | mogorman | thats right fubster |
| 05:09.43 | Fubster | so, that's not abnormal? |
| 05:09.51 | Math` | hey syle whats up |
| 05:09.59 | syle | hey man |
| 05:10.04 | mogorman | nope |
| 05:10.04 | X-Rob | Fubster, what is your audio problem? |
| 05:10.13 | syle | just chatting, avoiding my c programming as long as i can :) |
| 05:10.20 | Blankman | Qwell: go to blank_issue for a sec, I want to review something with :-) |
| 05:10.28 | Math` | lol that was my next question, how's your addon? |
| 05:10.37 | Fubster | X-Rob- it's just all-arount terrible. extremly choppy, even when i dial a local exten and try to get it to play back something. |
| 05:10.58 | X-Rob | sip phones? |
| 05:11.00 | syle | its doing everything it should now, i think the c program is around 2500 lines of code now, think i;ll finish it at 3000 with the new features |
| 05:11.01 | Fubster | this is with the standard prepackaged asterisk sounds |
| 05:11.02 | Lloydie-t | I think I might give up on the sqlite thing. I've got a headache, no sleep yet and got to go to work in an 2 hours in the UK |
| 05:11.06 | Fubster | X-Rob- i'm using x-ten |
| 05:11.23 | Fubster | but i called via IPKall thru FWD, and i have the same problem |
| 05:11.34 | Fubster | from my cell |
| 05:11.51 | syle | i been thinking how to do crap with time.h to say lets do this rate on a call from 6am to 8pm and this rate on a call from 10pm -6am, biggest thing i been thinking about lately |
| 05:12.16 | syle | maybe for or while loops to check time is in between |
| 05:12.18 | syle | not sure yet |
| 05:12.32 | X-Rob | Fubster, I just msg'ed you an echo test dialplan. Try using that |
| 05:12.41 | Fubster | will do |
| 05:12.43 | X-Rob | see if the 'You are about to start an echo test' is lossy |
| 05:13.08 | syle | other than that man, i been pretty drunk last 2 weeks and had a good time |
| 05:13.15 | Math` | haha good stuff |
| 05:13.20 | syle | u? |
| 05:13.21 | Fubster | alright, i will. thanks |
| 05:13.40 | X-Rob | no, do that now, and tell me 8) |
| 05:13.47 | X-Rob | this is diagnostics part 1 |
| 05:13.56 | Math` | I went raving on NYE, starting to do consulting for voip systems |
| 05:14.21 | syle | yeah same, i found a job other day online, sent me the NDA today to sign, see if i can get enough hours |
| 05:14.37 | Math` | cool |
| 05:14.55 | syle | work from home asterisk setups basically |
| 05:15.03 | Math` | voipsupply charged 480$US shipping fees for 11 polycom 301 |
| 05:15.28 | Fubster | X-Rob- it's still very slow and choppy |
| 05:15.29 | Math` | + we expect border fees |
| 05:15.39 | syle | omg |
| 05:15.47 | Math` | thats.... the last time I deal with them |
| 05:15.48 | syle | thats nuts, can;t they put them all in the same box |
| 05:15.58 | Math` | they should |
| 05:15.59 | X-Rob | Fubster, OK. This now means it's your PC or your network (or, the asterisk box is overloaded?) |
| 05:16.04 | Math` | williamsglobal is reseller in Canada tho |
| 05:16.08 | X-Rob | what distro and hardware? |
| 05:16.13 | X-Rob | (is your linux machine) |
| 05:16.22 | Math` | we'll stick to them, skipping border fees by the same occasion |
| 05:16.51 | Fubster | it's a p4, 128mb with an X environment running gentoo |
| 05:16.57 | X-Rob | what kernel? |
| 05:16.57 | Fubster | with the latest version of asterisk |
| 05:17.02 | X-Rob | 2.6....? |
| 05:17.06 | Fubster | yes |
| 05:17.09 | syle | border is just a pain to begin with, custom brokers, gst import numbers etc, if its not to much more money i usually buy from canada to |
| 05:17.15 | X-Rob | no, specifically, which version? |
| 05:17.20 | *** join/#asterisk sonic2wb (i=sonic2wb@user-11208d5.dsl.mindspring.com) |
| 05:17.26 | X-Rob | 2.6.what? |
| 05:17.29 | Fubster | <PROTECTED> |
| 05:17.46 | sonic2wb | Good Evening Everyohne |
| 05:17.50 | sonic2wb | Good Evening Everyone |
| 05:17.51 | X-Rob | 14, ok. Should be ok. However: Turn off X. |
| 05:17.56 | X-Rob | Problem 1. |
| 05:18.02 | Fubster | i'll try that |
| 05:18.15 | Fubster | let me see what top gives me before i do... |
| 05:18.16 | X-Rob | That _is_ a problem. X and Asterisk don't co-habit happily |
| 05:18.34 | X-Rob | S'not top, it's your video interrupts. |
| 05:18.39 | X-Rob | ctrl-alt-backspace |
| 05:18.45 | sonic2wb | init 3 |
| 05:18.46 | X-Rob | *kablam* |
| 05:18.48 | X-Rob | Or yea, init 3. |
| 05:18.52 | Fubster | i'm running this under a shell with screen |
| 05:19.03 | Fubster | so even though x is off, i'm still on |
| 05:19.12 | X-Rob | X is off? |
| 05:19.25 | X-Rob | ps auxww | grep X > dev/null && echo No it isnt |
| 05:19.26 | syle | i run asterisk in screen session to |
| 05:19.42 | X-Rob | wups |
| 05:19.45 | X-Rob | ps auxww | grep X > /dev/null && echo No it isnt |
| 05:20.11 | X-Rob | uh. And that won't work either. |
| 05:20.17 | X-Rob | ps auxww | grep -v grep | grep X > /dev/null && echo No it isnt |
| 05:20.22 | sonic2wb | there u go |
| 05:20.23 | X-Rob | There! Thats' actually accurate! |
| 05:20.24 | sonic2wb | lol |
| 05:20.26 | inv_Arp | pgrep x |
| 05:20.27 | syle | fubster maybe you can tell me how to reconnect to screen session upon boot, i use /usr/bin/screen -L -d -m /bin/nice -n -19 /usr/sbin/asterisk -U asterisk -vvvgc ...but usually i have to kill it and restart it in a screen session again so i can keep connecting to it |
| 05:20.52 | syle | i think it has no tty, maybe the reason not sure |
| 05:20.52 | sonic2wb | why are u guys running asterisk in a screen? |
| 05:21.09 | X-Rob | sonic2wb, if they have digium g729 licences, it needs a tty |
| 05:21.10 | inv_Arp | sonic2wb: I run th console in screen |
| 05:21.28 | Fubster | X-Rob- X is off |
| 05:21.29 | rob0 | I used to do that but now I just use -r when I want a console |
| 05:21.34 | Fubster | let me call with my cell |
| 05:21.38 | X-Rob | No |
| 05:21.39 | sonic2wb | right |
| 05:21.39 | X-Rob | nonono |
| 05:21.46 | X-Rob | You need to fix the local problems first |
| 05:21.50 | Fubster | syle- when you reboot your sreen sessions are lost |
| 05:21.59 | *** join/#asterisk __Soul__ (n=Soul@87-196-13-46.net.novis.pt) |
| 05:22.01 | Fubster | X-Rob- what do you mean? |
| 05:22.06 | X-Rob | You did an echo test |
| 05:22.16 | syle | i don;t think you understand what i said but ok :) |
| 05:22.16 | Fubster | right, while X was running |
| 05:22.28 | Fubster | syle- uh, maybe not... |
| 05:22.28 | X-Rob | that went from x-ten to asterisk and back again |
| 05:22.32 | X-Rob | that was broken |
| 05:22.37 | Fubster | right. |
| 05:22.39 | syle | this runs from rc.local |
| 05:22.40 | X-Rob | Do the echo test again without X |
| 05:23.31 | syle | i was wondering if maybe its as simple as executing a shell script with /bin/bash and starting screen, maybe i;d have a tty then |
| 05:23.52 | Math` | you need to run asterisk in a tty for 729? |
| 05:23.59 | Math` | why? |
| 05:24.34 | X-Rob | ask digium. |
| 05:24.39 | mogorman | ? |
| 05:24.40 | X-Rob | I'm sure it's a licencing thing. |
| 05:24.43 | mogorman | why are we asking digium? |
| 05:24.52 | sonic2wb | this navite bridgeing thing asterisk does does it not do it with sip clients? |
| 05:24.55 | Fubster | X-Rob- i called my IPKall with my cell phone |
| 05:24.56 | X-Rob | because it's their licence |
| 05:24.59 | Fubster | and it's still doing it |
| 05:25.09 | X-Rob | Fub, well that's totally useless, you know? |
| 05:25.10 | sonic2wb | fub: what the problem? |
| 05:25.26 | Fubster | X-Rob- what do you mean? |
| 05:25.32 | Fubster | it gave a warning: WARNING[14748]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame |
| 05:25.34 | X-Rob | We have the problem down to three things. X-ten, your network, or your asterisk box |
| 05:25.40 | X-Rob | why are you adding extra stuff into the mix? |
| 05:25.44 | sonic2wb | hm |
| 05:25.50 | ManxPowe | Fubster, that message is more or less harmless. |
| 05:25.55 | sonic2wb | thats what i say |
| 05:25.56 | sonic2wb | ? |
| 05:26.06 | Fubster | X-Rob- uh, it just calls my box via FWD |
| 05:26.14 | Fubster | i don't have a CLI softphone |
| 05:26.18 | X-Rob | Yes. But that's not broken. |
| 05:26.20 | X-Rob | What? |
| 05:26.23 | Fubster | unless you can dial from the asterisk CLI? |
| 05:26.35 | X-Rob | Ah. You don't have a windows box? |
| 05:26.39 | Fubster | nope |
| 05:26.41 | Fubster | this is it |
| 05:26.47 | sonic2wb | X-ten or SJPHONE |
| 05:27.02 | sonic2wb | ek sorry for caps |
| 05:27.08 | sonic2wb | shift key stuck |
| 05:27.11 | ManxPowe | Fubster, if you have a sound card installed and OSS or ALSA libs and headers installed when you build Asterisk, then the build process will enable the "dial" cli command |
| 05:27.29 | X-Rob | OK. So you had x-ten on the same box as the asterisk and it was still droppy-outty |
| 05:27.33 | X-Rob | something's fubar. |
| 05:27.37 | X-Rob | and I don't have time to figure it out. |
| 05:27.39 | syle | anyone want to buy a TDM400P card 2 fxs + 2 fxo modules, let me know , i;ll give you a good deal on it |
| 05:27.50 | ManxPowe | syle, no thanks!! |
| 05:27.55 | sonic2wb | syle pm me with price |
| 05:28.02 | sonic2wb | lol |
| 05:28.06 | Fubster | X-Rob- yeah |
| 05:28.14 | X-Rob | Go install CentOS 4.2 or something and I may be able to help, but you'll need to find someone with a working gentoo box so you can check revisions and stuff. |
| 05:28.21 | X-Rob | Sorry 8-\ |
| 05:28.27 | sonic2wb | Fub i can help im on gentoo |
| 05:28.48 | sonic2wb | lol |
| 05:28.49 | Fubster | i just did dial <exten>@context |
| 05:28.54 | Fubster | and it's still laggy as all hell |
| 05:28.58 | Fubster | this is really weird... |
| 05:29.16 | sonic2wb | fub: processor and speed? |
| 05:29.33 | Fubster | Intel(R) Pentium(R) 4 CPU 1.60GHz |
| 05:29.41 | sonic2wb | hmmz |
| 05:29.49 | Fubster | 128mb of ram... |
| 05:30.08 | sonic2wb | what voip service? |
| 05:30.25 | Fubster | none. i just use freeworlddialup |
| 05:31.15 | sonic2wb | are u behind a firewall? |
| 05:31.48 | Fubster | not on this machine, but i have a linksys WRT54G that's firewalled |
| 05:32.02 | Fubster | i'm doing all this on localhost though, so it shouldn't matter |
| 05:32.08 | Fubster | all ports are forwarded |
| 05:33.03 | ManxPowe | You need to do MUCH more than just forwarding ports if you want Asterisk to be behind NAT and use a SIP device (or server) that is outside the local network. |
| 05:33.06 | ManxPowe | See the Wiki. |
| 05:33.10 | ManxPowe | ~docs |
| 05:33.11 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 05:34.03 | sonic2wb | im going to say this but only from experince: SIP DONT WORK BEHIND FIREWAll EVEN if ports are fowarded, i just spent 2weeks trying to figure out my problem same as yours i moved my asterisk to outside of firewall and it works just fine |
| 05:34.27 | Fubster | your sound was all whacked up too? |
| 05:34.30 | sonic2wb | yes |
| 05:34.35 | Fubster | hmm, let me try something... |
| 05:34.39 | sonic2wb | people sound like robots? |
| 05:34.49 | sonic2wb | etc.... |
| 05:34.50 | Fubster | yeah, sort of |
| 05:34.50 | ManxPowe | CTRL-A DEL: The easy way to catch up on your asterisk-users mail. |
| 05:34.51 | syle | firewalls work fine if your default rule is allow |
| 05:34.58 | syle | then just deny the fags you want to |
| 05:35.18 | ManxPowe | sonic2wb, SIP works just fine behind firewalls and NAT. It's just rather more complicated to set up. |
| 05:35.45 | sonic2wb | Manx i have done everything everyone said on here and in websites it just wouldnt work |
| 05:35.52 | *** join/#asterisk linagee (n=linagee@netblock-68-183-1-55.dslextreme.com) |
| 05:36.00 | sonic2wb | put asterisk outside and it works like a charm |
| 05:36.41 | ManxPowe | sonic2wb, I have personally run, in production: Asterisk behind NAT/firewall with a SIPura ATA that worked on the same LAN, outside the LAN on a public IP, and outside the LAN, behind another NAT router. |
| 05:37.03 | syle | Protocol Ports Service |
| 05:37.03 | syle | UDP 5060 SIP |
| 05:37.03 | syle | UDP 5036 IAX |
| 05:37.03 | syle | UDP 4569 IAX2 |
| 05:37.03 | syle | UDP 20000 to 21000 RTP |
| 05:37.24 | sonic2wb | yep and for some ungodly reason mine wouldnt work |
| 05:37.29 | ManxPowe | It took me about 10 mins to do it, but 1) I know Asterisk, 2) I know RTP issues with NAT, 3) I know my router, 4) I know IP. |
| 05:37.31 | Fubster | well i just put myself in the DMZ and shut off my router firewall |
| 05:37.36 | Fubster | and it's still not working :\ |
| 05:37.39 | sonic2wb | asterisk sip server behind nat, sip clinet behind nat |
| 05:37.39 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 05:37.50 | syle | should work, maybe sure your ports are right in rtp.conf for your rtp settings |
| 05:37.51 | sonic2wb | fub reinstall * |
| 05:38.03 | Fubster | yeah i'm gonna do that. |
| 05:38.09 | Fubster | this is really weird |
| 05:38.26 | ManxPowe | HOWEVER, people that are just starting with VoIP should NOT run anything with NAT between devices. |
| 05:38.37 | ManxPowe | You have a steep enough learning curve with Asterisk. |
| 05:38.55 | sonic2wb | Asterisk isnt that hard if you can read your ABC's |
| 05:39.15 | Fubster | i get how it works and all, but just this audio problem... |
| 05:39.25 | Fubster | it's baffling me |
| 05:39.43 | ManxPowe | Fubster, you are not doing something only an idiot would do and have allow=all in sip.conf, do you? |
| 05:40.00 | sonic2wb | thats what i was typeing |
| 05:40.01 | sonic2wb | lol |
| 05:40.23 | ManxPowe | Top 10 Ways to Tell If Someone Is a Newbie: |
| 05:40.29 | ManxPowe | 1) use of "r" option to dial |
| 05:40.34 | ManxPowe | 2) allow=all |
| 05:40.45 | ManxPowe | 3) tT options to Dial |
| 05:40.45 | sonic2wb | r to dail? |
| 05:40.47 | Fubster | ManxPowe- nope |
| 05:40.48 | ManxPowe | any others |
| 05:41.06 | Fubster | no, nothing like that |
| 05:41.19 | Fubster | besides, i've tried through both SIP and IAX2 |
| 05:41.27 | ManxPowe | 4) What is safe_asterisk |
| 05:41.43 | sonic2wb | lol i was going to ask that |
| 05:41.43 | ManxPowe | 5) Asks "How to I start Asterisk on boot?" |
| 05:42.05 | ManxPowe | 6) "No Application MeetMe" |
| 05:42.27 | syle | what is a PRI |
| 05:42.39 | syle | who is digium |
| 05:42.47 | sonic2wb | right manx we get the point but thats why we all are here right to help each other |
| 05:43.00 | syle | what is a DID |
| 05:43.14 | X-Rob | 7) This is too hard, where do I buy the heroin around here? |
| 05:43.21 | Fubster | hey guys, where is my phone number? |
| 05:43.23 | Fubster | eeehehe |
| 05:43.26 | sonic2wb | lol |
| 05:43.29 | ManxPowe | X-Rob, I've asked that a time or two |
| 05:43.42 | Fubster | "too hard brb pot" |
| 05:44.16 | ManxPowe | Hmm..I've owned my car for 2 months and already put 4,000 miles on it. |
| 05:44.37 | sonic2wb | you must not drive much |
| 05:45.10 | ManxPowe | sonic2wb, I currently live 10 miles from the nearest gas station, on the top of a mountian. |
| 05:45.11 | Qwell | ManxPowe: 0) I'm using Asterisk@Home, and ... |
| 05:45.21 | sonic2wb | rofl |
| 05:45.34 | Qwell | That definitely trumps all the others. |
| 05:45.38 | ManxPowe | Also the texas <-> alabama runs eat up the miles, as do the alabama <-> louisiana runs |
| 05:45.38 | Qwell | except maybe tT |
| 05:45.39 | mogorman | qwell wtf? |
| 05:46.02 | sonic2wb | manx do u live where i do? |
| 05:46.03 | sonic2wb | lol |
| 05:46.08 | Qwell | ManxPowe: man, you drive like we do... |
| 05:46.21 | ManxPowe | Qwell, before 2 months ago I never owned a car. |
| 05:46.25 | mogorman | qwell when / why did you go to dark side |
| 05:46.33 | Fubster | wait, i just noticed something |
| 05:46.43 | Fubster | what exactly is "flexible rate"? |
| 05:46.46 | mogorman | or are you kidding |
| 05:46.53 | Fubster | might that have anything to do with it? |
| 05:46.54 | ManxPowe | Fubster, that's a message from mpg123 |
| 05:46.57 | ManxPowe | Fubster, no. |
| 05:47.03 | Fubster | poo |
| 05:47.31 | sonic2wb | fub whats ur netstat -a say about ports |
| 05:47.34 | ManxPowe | sonic2wb, I live on Chandler Mountian, about 50 miles NE of Birmingham AL |
| 05:48.07 | Qwell | ManxPowe: 8) Use of the default context |
| 05:48.13 | ManxPowe | It's pretty nice up on the mountian |
| 05:48.36 | Fubster | sonic2wb- it doesn't mention asterisk |
| 05:48.39 | Fubster | or sip, or IAX... |
| 05:48.46 | ManxPowe | Qwell, we should actually write up a Top 10 List for this and post it to the mailinglists. |
| 05:48.51 | mogorman | gnite |
| 05:49.16 | Qwell | ManxPowe: I'm game |
| 05:49.34 | Qwell | I like your 1, 3, 6 at least |
| 05:49.41 | ManxPowe | Qwell, I would have to care enough to make time. |
| 05:50.00 | ManxPowe | GADS having real broadband again is nice, even if its only for afew days. |
| 05:50.07 | Fubster | whatver i'll work on this tomorrow. night all |
| 05:50.27 | sonic2wb | MAnx: ah a southerner i live in West NC |
| 05:50.48 | ManxPowe | sonic2wb, I'm a yankee, but have lived in the south for 13 or 14 years. |
| 05:50.55 | Fubster | weird thing is, when i call my softphone from my cell phone, it gets a decent signal |
| 05:50.57 | ManxPowe | Lived on the mississippi gulf coast until recently. |
| 05:51.06 | Fubster | not the best, but not as bad as the played sounds |
| 05:51.53 | _Soul_ | Qwell, thanks 4 all the help. the phone does not register with our asterisk, tought in the configuration i told him to |
| 05:51.55 | Qwell | ooo, I got another one |
| 05:52.00 | Qwell | ManxPowe: 9) _. |
| 05:52.17 | _Soul_ | the sip configuration is correct, but "register with proxy" is off, is this normal ? |
| 05:52.34 | ManxPowe | Qwell, I'm about 40 miles north of New Orleans at the moment (Hammond, LA) |
| 05:52.58 | ManxPowe | On a link with less than 800ms latency! |
| 05:53.37 | Qwell | not too shabby |
| 05:53.55 | *** join/#asterisk king_elephant (n=user@ool-43516159.dyn.optonline.net) |
| 05:54.09 | Qwell | 10) I bought an x100p off ebay, and... |
| 05:54.58 | *** part/#asterisk king_elephant (n=user@ool-43516159.dyn.optonline.net) |
| 05:55.55 | sonic2wb | 11) What is linux |
| 05:55.55 | ManxPowe | Qwell, 11) I can't spend any money..... |
| 05:56.02 | sonic2wb | lol |
| 05:56.18 | Qwell | 11b) So I bought a gxp2000 |
| 05:56.29 | ManxPowe | Qwell, or any grandstream product. |
| 05:56.32 | Qwell | indeed |
| 05:56.44 | ManxPowe | This $50 phone I bought sucks! I feel so ripped off! |
| 05:56.56 | Blankman | You guys are not nice ... ;-) |
| 05:57.05 | Qwell | hey, it's true :p |
| 05:57.12 | sonic2wb | im going to ask probilly a dumb question where can i get one of those intel chipset modems to hack to use as a fXO card? |
| 05:57.21 | Qwell | sonic2wb: see 10) |
| 05:57.29 | sonic2wb | funny |
| 05:57.30 | Blankman | Ebay? |
| 05:57.32 | infinity1 | ahhah ebay |
| 05:57.32 | Qwell | not really :p |
| 05:57.39 | sonic2wb | not ebay |
| 05:57.46 | Qwell | sonic2wb: That's about the only place |
| 05:57.47 | infinity1 | just get a 400p |
| 05:57.58 | sonic2wb | ebay = worse than satan |
| 05:58.14 | syle | someone told me other day amazon was better |
| 05:58.16 | Blankman | You can get a clone for less than you will pay for the modem :-) |
| 05:58.17 | syle | whatever that means |
| 05:59.06 | ManxPowe | The people that deserve not sympathy from me are the ones that go out and buy 50 Grandstream phones and deploy their first Asterisk server with them and then are amazed it doesn't work well. They seem to get all miffed when I tell them they should have 1) read the mailing list posts about the products they are considering 2) order the top three make/models of phones they are considering, 3) deploy a TEST System and use inte |
| 05:59.07 | ManxPowe | rnal beta testers |
| 05:59.28 | sonic2wb | whats the chances of someone writeing code for other chipsets? |
| 05:59.36 | Qwell | I'm SO glad we have money to put behind our deployment |
| 05:59.38 | ManxPowe | sonic2wb, nobody with the skills care. |
| 05:59.47 | sonic2wb | thats true |
| 05:59.47 | infinity1 | ManxPowe: wow. thats big screw up. |
| 06:00.22 | infinity1 | ManxPowe: i think what happens is people buy one gxp to test with and end up giving to their most hated user |
| 06:00.28 | ManxPowe | Qwell, We tested Grandstream (just to see if they were as bad as everyone says -- they are), Zultys SIP2, Cisco 7905, Polycom IP 300, Cisco ATA, Sipura 841, Sipura ATA |
| 06:00.44 | infinity1 | ManxPowe: no snom? |
| 06:00.49 | ManxPowe | infinity1, no. |
| 06:00.52 | Qwell | well, we're going with mostly 7960's, so...that will be very easy |
| 06:00.58 | Qwell | 796x |
| 06:01.00 | ManxPowe | infinity1, there were too many reports of firmware issues. |
| 06:01.04 | syle | whats wrong with polycom? |
| 06:01.14 | Qwell | syle: nothing, polycom is great |
| 06:01.17 | infinity1 | ManxPowe: what did you go with? |
| 06:01.17 | ManxPowe | We went with Polycom for phones, SIPura for ATAs |
| 06:01.24 | Blankman | ManxPowe: you do an iax testing? |
| 06:01.30 | Qwell | ManxPowe: Have you tried the SPA941? |
| 06:01.32 | sonic2wb | anyways good talking with everyone THanks Qwell for the advise, i may be back if i cant sleep |
| 06:01.32 | syle | he said polycom 300 suck |
| 06:01.37 | ManxPowe | Blankman, Hmm? |
| 06:01.50 | ManxPowe | Qwell, no, as I said we standardized on Polycom. |
| 06:01.54 | Qwell | syle: he said the GS suck |
| 06:01.56 | infinity1 | ManxPowe: i'm not sure if our next phone will be another polycom or a snom. i heard some good stuff about snom |
| 06:02.23 | SLiCKFX | y3llo |
| 06:02.23 | syle | cisco |
| 06:02.30 | {zombie} | the really good thing about snom is that if you have an issue with their firmware it's fixed within days |
| 06:02.31 | ManxPowe | Our first production Asterisk server went live 25 months ago. |
| 06:02.35 | syle | hmmmm |
| 06:02.36 | Blankman | Just wondering ... we have found anything better than the IAXy ... yet :-) |
| 06:02.41 | syle | i don;t like their licensing |
| 06:02.42 | ManxPowe | We had about 6 - 9 months of testing before deloyment. |
| 06:02.54 | ManxPowe | Blankman, Oh, we tried the IAXy too. |
| 06:03.04 | syle | how was that? |
| 06:03.23 | Blankman | ManxPowe: the old or new ones? |
| 06:03.24 | ManxPowe | The Cisco and Polycom were the phones we liked and worked will for us. |
| 06:03.51 | ManxPowe | Blankman, like the 1st production run, but I feel the DESIGN of the IAXy is lacking. |
| 06:03.57 | Qwell | ManxPowe: If you ever do another rollout, you should revisit the cisco's with sccp... |
| 06:04.09 | ManxPowe | Qwell, Sticking with Polycoms. |
| 06:04.12 | SLiCKFX | sup all what is a route considered... is it unlimited IAX termination to the specified area? |
| 06:04.21 | ManxPowe | I'm a big believer in picking a vendor and sticking with it. |
| 06:04.32 | syle | blah useless sip phones again, what have you tested as far as cordless/wifi sip phones, useful ones where you can walk around with them wirelessly |
| 06:04.51 | ManxPowe | SLiCKFX, Please speak english. |
| 06:04.53 | SLiCKFX | Im looking for steady access to Taiwan proper + cellular |
| 06:04.56 | _Soul_ | Qwell, thanks 4 all the help. the phone does not register with our asterisk |
| 06:05.12 | Qwell | _Soul_: That's what the SIPmac.cnf is for |
| 06:05.13 | Blankman | ManxPowe: I don't disagree ... we will have an ATA that we are building out in Q4 we hope ... still waiting to figure out how we want to do the encryption though ... |
| 06:05.13 | SLiCKFX | i see people asking for routes to certain places |
| 06:05.25 | ManxPowe | syle, We expect to test our first SIP Wifi phone in about 3 years. |
| 06:05.31 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 06:05.40 | _Soul_ | unlocked the settings menu, and filled the values ok, everything is correctly configured, but the phone does not reach the asterisk server |
| 06:05.48 | ManxPowe | SLiCKFX, mostly they ask on the asterisk-biz mailing lists, not here. |
| 06:05.50 | infinity1 | ManxPowe: guess you won't be testing one then :) probably a good idea |
| 06:05.57 | syle | 3 years? i;ve seen about 6 on the market now, anyone played with any? |
| 06:06.11 | Qwell | syle: I know twisted has one. I think the Hitachi |
| 06:06.14 | ManxPowe | syle, I've yet to hear much good about any of them. |
| 06:06.18 | SLiCKFX | i know i was wondering if a route is usuable with IAX/Asterisk |
| 06:06.30 | SLiCKFX | and wondering exactly what a route consists of |
| 06:06.53 | ManxPowe | SLiCKFX, usually it means "can a carrier send telephone calls to place X" |
| 06:07.30 | SLiCKFX | are there alot of people providing termination other than sites |
| 06:07.43 | syle | tons |
| 06:07.53 | syle | alot don;t have websites |
| 06:07.53 | SLiCKFX | coo |
| 06:08.00 | Blankman | nite all. |
| 06:08.06 | *** part/#asterisk Blankman (n=blankman@c-24-61-183-130.hsd1.nh.comcast.net) |
| 06:08.19 | syle | alot prefer the old fashioned way, call them :) |
| 06:09.22 | SLiCKFX | thx |
| 06:09.38 | sonic2wb | http://www.zyxel.com/product/P2000W.php |
| 06:10.21 | syle | i'd be a good tester for that phone |
| 06:10.29 | sonic2wb | http://www.zyxel.com/product/P2000W.php <-- wifi phone i dont know about thier phones but i use thier dsl equipment everyday and its very good |
| 06:10.35 | syle | IEEE 802.11b , i got a router for it, and xbox360 running |
| 06:10.39 | syle | check interference |
| 06:11.38 | syle | ever since i switched my router/wireless/switches over to dlink i haven;t had one problem |
| 06:11.55 | syle | i;ve had problems with netgear and linksys for inteference and packetloss on wireless |
| 06:11.59 | sonic2wb | hmm |
| 06:12.12 | sonic2wb | im haveing some probs with netgear atm too |
| 06:12.45 | syle | the netgear one i had to reset all the time |
| 06:12.49 | syle | was a pain |
| 06:13.24 | syle | even ripped it open and install custom cooling system in it |
| 06:13.26 | syle | still no go |
| 06:14.05 | sonic2wb | ack |
| 06:14.10 | sonic2wb | power is trying to go out |
| 06:14.15 | syle | i use the dlink gamingrouter |
| 06:14.17 | syle | its called |
| 06:14.24 | syle | and use it as an accesspoint |
| 06:14.32 | sonic2wb | ah |
| 06:15.04 | sonic2wb | DAM IT |
| 06:15.18 | sonic2wb | someone must of hit a powerpole |
| 06:15.24 | syle | none of my laptops have a problem with it, and they still ahve the wireless netgear G cards in them |
| 06:15.30 | sonic2wb | hmmz |
| 06:15.54 | sonic2wb | im useing a dlink G card with a netgear G router and i get droped packets etc.. |
| 06:16.46 | syle | yeah well get a dlink gamingrouter and see if you still get them, otherwise i;d look for inteference, but i think its your netgear like i had |
| 06:18.07 | sonic2wb | some nice wifi phones |
| 06:18.08 | sonic2wb | http://www.voipsupply.com/index.php?cPath=95_115&ref=google_wifi |
| 06:18.39 | syle | yeah sip phones won;t gain popularity until they can replace cordless phones |
| 06:18.54 | syle | who the hell wants to be wired to their desk on a call |
| 06:19.05 | syle | hell i piss and talk to people lol |
| 06:19.14 | sonic2wb | lol |
| 06:19.19 | sonic2wb | ive done worse |
| 06:19.28 | sonic2wb | i love my bluetooth headset |
| 06:19.29 | sonic2wb | lol |
| 06:19.58 | syle | well if your exiting the other way i usually hit mute for that groan lol |
| 06:19.59 | *** join/#asterisk chapeaurouge (n=chapeaur@85.201.81.201) |
| 06:20.47 | syle | one kewl wired sip phone i did see was from polycom |
| 06:20.56 | syle | was like 1k for a speaker phone |
| 06:21.04 | syle | looked pretty nice to |
| 06:21.10 | sonic2wb | i like this |
| 06:21.12 | sonic2wb | http://www.voipsupply.com/product_info.php?products_id=1067 |
| 06:21.16 | sonic2wb | looks like a cellphone |
| 06:21.43 | Math` | nice one |
| 06:22.18 | Math` | is there any GSM/CDMA + 802.11-SIP phones? |
| 06:22.25 | Qwell | sonic2wb: now if only it worked well... |
| 06:22.30 | syle | 300 bucks eeks, i think you can buy a regular cell phone, a gsm adapter for asterisk and switch out your sim cards , prob make more sense |
| 06:22.41 | Math` | syle: lol |
| 06:22.44 | Qwell | Math`: I don't know if any GSM providers would provision a phone they didn't sell you |
| 06:22.48 | Qwell | CDMA, sure |
| 06:22.53 | Qwell | or, other way around? :P |
| 06:22.54 | Math` | syle: how much is a gsm adapter for asteisk? |
| 06:22.57 | Qwell | which one has the SIMs again? |
| 06:23.00 | Math` | Qwell: GSM has SIMs :P |
| 06:23.03 | Qwell | right |
| 06:23.08 | syle | i think they were like 100-300 bucks |
| 06:23.09 | Qwell | what I said, reversed. heh |
| 06:23.14 | Math` | lol |
| 06:23.15 | syle | just need a sim unlocked phone |
| 06:23.21 | Math` | you probably can provision it yourself |
| 06:23.36 | Math` | take the "free" phone they give you and get the subscriber ID out of it |
| 06:23.41 | SLiCKFX | damn i got a SIP phone setup .. I can dial an ext. good but I thought having a "exten => _1NXXNXXXXXX,1,Dial" channel set in extensions would enable dialing 1 + area and number too call out normally... can someone help me out |
| 06:23.47 | Qwell | Math`: yeah, maybe |
| 06:23.48 | Math` | now HOW to do it is the next storyu |
| 06:24.06 | syle | here i will find link for you |
| 06:24.18 | syle | http://www.voip-info.org/tiki-index.php?page=How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX |
| 06:24.21 | syle | here ya go |
| 06:24.22 | Math` | syle: 300$ for a GSM gateway? |
| 06:24.34 | Math` | arent they 1k$+? |
| 06:25.44 | syle | i know a few people in here implemented this |
| 06:25.55 | syle | they switch out their sim cards when they get home |
| 06:26.15 | *** join/#asterisk g0mb0 (n=test@external.micom.mng.net) |
| 06:26.43 | syle | the future will be cell phones with voip capability for sure |
| 06:27.07 | syle | just depends on pricing as usual lol |
| 06:28.28 | Math` | syle: can't you program the gsm gateway to use your gsm provider's sim? |
| 06:28.52 | OloBola | is it possible to route an incoming voip call through my x100p (to a local number). So a person call through my provider (nufone) then asterisk calls a local number and puts the caller through |
| 06:30.11 | Math` | is the x100p an fxo card? |
| 06:30.25 | OloBola | yes |
| 06:30.31 | Math` | then you can do it |
| 06:30.38 | OloBola | great, thanks |
| 06:33.24 | SLiCKFX | OloBola how are u trying to route people through your server... phone card style or just call forwarding |
| 06:33.45 | SLiCKFX | to a set number |
| 06:35.14 | syle | i am not sure math, if i get one i;ll let you know :) |
| 06:35.28 | syle | i don;t have a cell phone or i would |
| 06:36.05 | OloBola | SLiCKFX: I just got my drivers working with my x100p. That is as far as I've gotten. I |
| 06:36.19 | syle | i;d probably get one of those sony erikson triband gsm phones unlocked if i did |
| 06:36.36 | Qwell | what is multiband? |
| 06:36.45 | Qwell | I obviously don't know cell tech :p |
| 06:36.51 | syle | you want triband |
| 06:36.59 | syle | or if your in the US it won;t work in europe |
| 06:37.00 | Qwell | I've see dual, tri, I think quad |
| 06:37.10 | syle | theres no quad |
| 06:37.17 | Qwell | maybe not then :p |
| 06:37.24 | Qwell | What's it mean though? |
| 06:37.46 | syle | what most people do is they fly to europe, when they get off the plane they switch out their sim card with a european providers card |
| 06:38.03 | syle | and vice-versa |
| 06:38.25 | Qwell | what does that have to do with the bandedness? |
| 06:38.27 | syle | unless your rich then who cares pay more |
| 06:38.32 | *** join/#asterisk [-- (i=somjuk@jane.lru.ac.th) |
| 06:38.34 | Qwell | (yes, I made that word up) |
| 06:38.59 | Math` | Qwell: GSM can work on multiple frequency sets, thease are called bands |
| 06:39.01 | syle | you have to look at the history of gsm to know what they are |
| 06:39.15 | Qwell | So, thats only for gsm phones? |
| 06:39.33 | syle | yeah |
| 06:39.50 | syle | i beleive sprint has been slowly switching over their network to the 3rd band like europes |
| 06:39.56 | Qwell | now, what about modes? |
| 06:40.01 | _Soul_ | err, anybody can help with this nasty 7940 ? the phone does not register itself with asterisk |
| 06:40.03 | Qwell | I recall seeing "dualmode triband" |
| 06:40.21 | _Soul_ | asterisk is working fine, i have about 100 xlite's registered there |
| 06:40.59 | syle | have you tried the example config from voipinfo for that phone? |
| 06:41.03 | _Soul_ | the 7940 SIP(mac).cnf was filled with the asterisk server's details, sip login and password |
| 06:41.25 | _Soul_ | whe i go to the 7940 sip menu, all the settings are correct |
| 06:41.32 | _Soul_ | but the phone does not register |
| 06:42.01 | syle | you sure? |
| 06:42.05 | syle | you checked debug logs? |
| 06:42.19 | syle | you check ethernet cord works etc? |
| 06:42.26 | Math` | _Soul_: you tried sniffing traffic? |
| 06:42.38 | _Soul_ | tailing /var/log/asterisk/full |
| 06:42.40 | OloBola | SLiCKFX: Call forwarding! I re-read your question, silly me. |
| 06:42.45 | syle | won;t have to, debug logs will show failed registered attempts |
| 06:42.46 | _Soul_ | Math`, err, no |
| 06:43.06 | _Soul_ | syle, yes, networking is fine, as i was able to upgrade its firmware |
| 06:43.30 | _Soul_ | the 7940 allows several sip proxies: |
| 06:43.43 | _Soul_ | unprovisioned proxy: backup |
| 06:43.50 | _Soul_ | unprovisioned proxy: emergency |
| 06:43.58 | syle | hehe |
| 06:44.02 | syle | thats kewl |
| 06:44.26 | _Soul_ | but i only have one asterisk server ;) |
| 06:44.26 | SLiCKFX | OloBola it should be as east as setting exten => zap,1,Answer zap,2,DIAL |
| 06:45.13 | Math` | SLiCKFX: exten should be s, not zap |
| 06:46.11 | syle | you mean tail /var/log/asterisk/debug |
| 06:46.19 | syle | if you have it enabled in logger.conf |
| 06:46.25 | _Soul_ | err, i was tailing full |
| 06:46.26 | SLiCKFX | coo yeah ive never messed with those cards |
| 06:47.15 | SLiCKFX | OloBola: got to register 1 sec. |
| 06:49.54 | _Soul_ | any ideias ? |
| 07:01.50 | xbmodder | my sipura is telling my asterisk box off: |
| 07:01.51 | xbmodder | Hello, |
| 07:01.51 | xbmodder | You have two OneLink lines both still under contract. One contract expires 03/07/2006 and the second expires 05/16/2006. |
| 07:01.51 | xbmodder | The OneLink installation can take between 15 to 20 Business Days. If you're ready for the move, please give our Sales Team a call at 800.556.5829. |
| 07:01.52 | xbmodder | I hope this information is helpful to you. I will be placing this ticket into a pending customer response state. It will close automatically in 48 hours if you don?t have any additional questions. |
| 07:01.55 | xbmodder | Thanks, |
| 07:01.57 | xbmodder | Mark J |
| 07:01.59 | xbmodder | Overnight Support |
| 07:02.01 | xbmodder | http://www.speakeasy.net/myspeak |
| 07:02.03 | xbmodder | Hours > 6pm - 5am PT Sat - Tues |
| 07:02.05 | xbmodder | oops |
| 07:02.07 | xbmodder | crap wrong e-mail |
| 07:02.09 | xbmodder | Jan 2 23:55:17 10.0.0.210 SIP/2.0 401 Unauthorized^M Via: SIP/2.0/UDP 10.0.0.210:5060;branch=z9hG4bK-4e5e44a0; |
| 07:02.12 | xbmodder | received=10.0.0.210^M From: sipura <sip:sipura@10.0.0.1>;tag=fbc93c0d89a3e2cco0^M To: sipura <sip:sipura@10.0.0 |
| 07:02.19 | xbmodder | .1>;tag=as342f2931^M Call-ID: 949b8bd1-3a3e35d8@10.0.0.210^M CSeq: 805 REGISTER^M User-Agent: Asterisk PBX^M Al |
| 07:02.24 | xbmodder | low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Max-Forwards: 70^M Contact: <sip:sipura@10.0 |
| 07:02.27 | xbmodder | .0.1>^M WWW-Authenticate: Digest realm="asterisk", nonce="1feee640"^M Content-Length: 0^M ^M |
| 07:02.28 | Qwell | ~pb |
| 07:02.32 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 07:02.42 | xbmodder | Qwell, sorry |
| 07:04.13 | *** join/#asterisk SLiCKFX (n=askme@pcp03218165pcs.hlcrs201.al.comcast.net) |
| 07:05.35 | xbmodder | http://pastebin.com/488340 < log from SIPURA > asterisk |
| 07:06.50 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-70-93-239-175.san.res.rr.com) |
| 07:09.28 | *** join/#asterisk koperniqs (n=koperniq@129.187.15.40) |
| 07:09.30 | koperniqs | hi |
| 07:15.03 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 07:15.26 | *** join/#asterisk vinko (n=zic@69.88.69.250) |
| 07:15.27 | elvisthedj | can anyone tell me if it's possible to do a 3 way call from the console? |
| 07:15.30 | elvisthedj | (using alsa) |
| 07:16.03 | drumkilla | elvisthedj: using MeetMe, yes. |
| 07:16.28 | elvisthedj | well, that won't work in my situation because i need dtmfs after the third party is called for an automated system |
| 07:16.36 | elvisthedj | the dtmfs are supressed by meetme |
| 07:17.09 | *** join/#asterisk woodchuck (n=woodchuc@S0106000000da2a3d.ok.shawcable.net) |
| 07:17.24 | *** join/#asterisk Katty (n=angela@68-112-15-110.dhcp.cpgr.mo.charter.com) |
| 07:17.51 | elvisthedj | so, there is no command or series of commands on the cli that will let me make the call? |
| 07:20.13 | Math` | you could call the third party, send the dtmf, then transfer the call to the meetme conference |
| 07:21.23 | elvisthedj | thx. that's a no go because it's not just a sequence. it's an automated system that needs lots of input |
| 07:21.54 | elvisthedj | i saw the bit of code in meetme that removes the dtmfs and tried to get rid of it, but no change.. the system still doesn't hear the digits |
| 07:22.04 | elvisthedj | this wouldn't be a problem if my iaxy hadn't died :( |
| 07:23.37 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:23.52 | *** join/#asterisk lehel (n=ddd@82.79.20.17) |
| 07:24.00 | elvisthedj | if anybody happens to get an idea, feel free to contribute :) i'm still googling, but apparently the console isn't a popular method of making calls |
| 07:24.05 | lehel | hello.. and, happy new year |
| 07:25.21 | elvisthedj | lehel: hi lehel. happy new year to you too :) |
| 07:25.32 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 07:28.02 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 07:32.48 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-51.claranet.co.uk) |
| 07:35.21 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:37.27 | *** part/#asterisk secure75 (n=mic@p549A2E0F.dip0.t-ipconnect.de) |
| 07:37.43 | iPBX | anyone know a softphone that runs on PS2 or XBOX? |
| 07:37.55 | *** join/#asterisk pakipenguin (n=Administ@linuxpakistan/admin/pakipenguin) |
| 07:38.00 | pakipenguin | good evening |
| 07:43.54 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 07:46.42 | syle | mythtv as a frontend on xbox has a softphone |
| 07:48.25 | syle | new colo's, rackable xbox360's hehe |
| 07:48.33 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 07:48.42 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 07:49.20 | NoRemorse | hi, does psgw run on linux or [expletive] |
| 07:54.18 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 08:10.26 | *** join/#asterisk fugitivo (n=ajf@201.255.176.233) |
| 08:14.51 | *** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com) |
| 08:15.30 | jonathh | morning gentleman |
| 08:20.07 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 08:27.32 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:28.02 | iPBX | ZOMG Windows Rules! |
| 08:29.01 | iPBX | jk lol! |
| 08:29.23 | iPBX | So I could use an Xbox 360 as a Sip termination! |
| 08:29.24 | iPBX | sweet |
| 08:29.51 | xbmodder_lappy | iPBX, umm |
| 08:29.53 | xbmodder_lappy | XBOX1 yeah |
| 08:29.57 | xbmodder_lappy | xbox2, no |
| 08:30.00 | iPBX | :-( |
| 08:30.43 | *** join/#asterisk secure75 (n=mic@ppp-82-135-0-18.mnet-online.de) |
| 08:30.54 | iPBX | wonder if a video softphone application could be made for xbox... or is there one already? |
| 08:31.21 | *** join/#asterisk indego (n=chris@floyd.gms.lu) |
| 08:32.14 | syle | read up |
| 08:32.29 | iPBX | yea, i see softphone... I'm talking video softphone |
| 08:32.35 | webmind | haven't heard of it.. try google.. would be cool |
| 08:32.45 | iPBX | xbox have webcams for it? |
| 08:32.45 | webmind | :) |
| 08:32.54 | webmind | xbox1 has usb hasn't it ? |
| 08:33.00 | iPBX | i dunno |
| 08:33.01 | iPBX | i don't have one |
| 08:33.03 | webmind | afaik |
| 08:33.10 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:33.10 | iPBX | lol |
| 08:34.16 | iPBX | my sister has one, and my brother. I have an asterisk server, if it was possible to setup their xbox's to use it, they'd uh... shit there pants |
| 08:34.38 | iPBX | i think that's all they do is play xbox |
| 08:35.06 | iPBX | ~weather kmwn |
| 08:36.45 | webmind | iPBX, well try and get linux on it first.. shouldn't be a problem.. then you can connect devices to it and run asterisk on it |
| 08:40.14 | X-Rob | webmind, xbox1's controllers are USB with a non-standard connector. You can buy adaptors for about $2 |
| 08:40.53 | webmind | iPBX, there you go.. usb support.. therefor usb webcam support :) |
| 08:42.32 | X-Rob | note, no rtc circutry in xbox, so you have to use usb ztdummy (blech) |
| 08:42.33 | OloBola | so.. exten => s,1,wait(1) |
| 08:42.33 | OloBola | exten => s,2,Answer |
| 08:43.04 | OloBola | should pickup my x100p line? I can dial out now, but it doesn't seem to pickup the incoming |
| 08:51.33 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 08:53.23 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:57.26 | *** join/#asterisk Dibbler_ (n=Dibbler@zidane.pi-net.net) |
| 09:00.44 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-70-93-239-175.san.res.rr.com) |
| 09:05.15 | OloBola | d |
| 09:12.43 | *** join/#asterisk Darkhalf_ (n=darkhalf@cpe-70-93-239-175.san.res.rr.com) |
| 09:15.21 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:19.31 | *** join/#asterisk venix (n=venix@CPE000625f5bd17-CM0011aea5246c.cpe.net.cable.rogers.com) |
| 09:23.34 | *** join/#asterisk Blackmore41 (n=Miranda@p62.246.67.146.tisdip.tiscali.de) |
| 09:23.49 | Blackmore41 | hi |
| 09:24.14 | *** join/#asterisk Neal` (n=LostShad@adsl-67-120-234-12.dsl.sndg02.pacbell.net) |
| 09:25.00 | Blackmore41 | hi neal |
| 09:25.10 | Neal` | Hi (2) |
| 09:26.19 | Neal` | Heh. |
| 09:26.32 | *** part/#asterisk Neal` (n=LostShad@adsl-67-120-234-12.dsl.sndg02.pacbell.net) |
| 09:28.15 | *** join/#asterisk RaYmAn-B1 (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 09:36.37 | *** join/#asterisk fulgas (n=fulgas@213.58.130.46) |
| 09:47.01 | *** join/#asterisk lorinc (n=ang@caracas-0117.adsl.interware.hu) |
| 09:50.04 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:50.22 | *** join/#asterisk cianhughes (n=cian@87.192.36.98) |
| 09:52.30 | *** join/#asterisk laura (n=laura@host108-155.pool8252.interbusiness.it) |
| 09:52.47 | *** part/#asterisk laura (n=laura@host108-155.pool8252.interbusiness.it) |
| 09:54.15 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:56.21 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 10:00.06 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 10:02.26 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:02.52 | *** join/#asterisk Hmmmm (n=Hmmmm@221.135.51.19) |
| 10:11.30 | *** join/#asterisk emrah_ (n=emrah@knsrv1-zrh8048.net1.kavun.ch) |
| 10:11.53 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:12.02 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 10:17.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 10:19.28 | iPBX | ok here's my bizarre question for the day. can mpg123 play an mp3 stream? |
| 10:19.55 | iPBX | if so, how can i configure musiconhold.conf to play an mp3 stream |
| 10:19.58 | iPBX | instead of files |
| 10:30.05 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 10:35.59 | *** join/#asterisk svenna_ (n=svenna@p548D3422.dip0.t-ipconnect.de) |
| 10:36.14 | *** part/#asterisk svenna_ (n=svenna@p548D3422.dip0.t-ipconnect.de) |
| 10:40.36 | *** join/#asterisk sunil (n=sunil@202.54.37.186) |
| 10:52.35 | pr0m | ipbx: search 'musiconhold' on voip-info.org. |
| 10:52.44 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 10:53.35 | RoyK | wtf is the -dev list being censored? |
| 10:58.43 | tzafrir_laptop | RoyK, have you verified you write from a subscribed address? |
| 10:58.50 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:59.03 | OloBola | can I paypal someone 20 bucks or so to get my softphone to dial out through my x100p? The card is installed and should work. |
| 10:59.11 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 10:59.31 | OloBola | I'm just lost with this and tired of screwing around with it |
| 10:59.56 | RoyK | tzafrir_laptop: yes, several times |
| 11:00.00 | Mimmus | "Don't know what to do if second ROSE component is of type 0x6" |
| 11:00.00 | Mimmus | "!! Unknown IE 26 (cs6, Unknown Information Element)" |
| 11:00.00 | Mimmus | What are these? |
| 11:00.14 | Mimmus | I'm having these messages for every call |
| 11:00.37 | RoyK | tzafrir_laptop: yesterday, wrote from a subscribed address, roy@karlsbakk.net, got held by moderator, re-registered with roy@briiz.no, got through, then, after a few hours, i did not get through anymore |
| 11:00.59 | RoyK | tzafrir_laptop: so i re-registered now with a new address and sent a small message about censorship |
| 11:02.21 | chapeaurouge | are there any phone like the polycom ip 500, but wireless? |
| 11:02.53 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 11:03.06 | ramtha | hi, is there a working AOC support in asterisk? |
| 11:03.45 | X10ZION | anyone used Loquendo with Asterisk? |
| 11:04.01 | tzafrir_laptop | RoyK, got my email? |
| 11:04.38 | RoyK | answered with three emails |
| 11:05.10 | RoyK | tzafrir_laptop: currently only roy@karlsbakk.net is registered with -dev |
| 11:05.20 | RoyK | tzafrir_laptop: I can receive messages fine, but I can not post |
| 11:06.46 | tzafrir_laptop | RoyK, before posting such things to the whole list, have you tried contacting the moderator? |
| 11:07.06 | RoyK | yes |
| 11:07.24 | tzafrir_laptop | And ? |
| 11:07.28 | RoyK | no answer |
| 11:08.21 | tzafrir_laptop | It would have looked better if you mentioned this fact in your post. That you don't waste the list reader's time before you tried simpler means. |
| 11:09.14 | RoyK | sorry |
| 11:09.16 | RoyK | i forgot |
| 11:09.31 | RoyK | still getting censored like that pisses me off |
| 11:09.50 | RoyK | i'm trying to something as rotten as sharing some software I've ported to 1.2 |
| 11:12.51 | zoa | roy |
| 11:12.57 | zoa | you are NOT getting censored |
| 11:13.10 | zoa | i see all those emails |
| 11:13.19 | zoa | there is no moderator |
| 11:13.58 | iDunno | maybe his spam filter caught it ;) |
| 11:14.03 | zoa | if a message does not go through, its because you used the wrong sender address |
| 11:21.37 | Mimmus | good morning, does anyone know the meaning of this message: |
| 11:21.40 | Mimmus | "!! Unknown IE 26 (cs6, Unknown Information Element)"? |
| 11:22.07 | RoyK | zoa: strange thing those emails are not in the archives |
| 11:22.07 | Mimmus | peraphs something related to PRI signalling in Italy and not correctly implemented in libpri? |
| 11:22.14 | *** join/#asterisk xianlp (n=xian_1@193.170.41.114) |
| 11:22.25 | RoyK | zoa: http://lists.digium.com/pipermail/asterisk-dev/2006-January/subject.html |
| 11:22.37 | RoyK | zoa: you see, the messages from my comes from different email addresses |
| 11:24.17 | xianlp | hi there |
| 11:24.42 | XIN01OZ | where? |
| 11:24.48 | RoyK | zoa: ? |
| 11:27.37 | *** join/#asterisk amir (n=amir@gentoo/developer/amir) |
| 11:28.25 | *** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62) |
| 11:28.44 | nfi|ermes | i can t login in the mailbox |
| 11:29.14 | xianlp | me too :) |
| 11:29.34 | nfi|ermes | *98,3,VoiceMailMain(default) |
| 11:29.38 | XIN01OZ | anyone here used Loquendo with Asterisk? |
| 11:29.41 | nfi|ermes | it says incorrect password |
| 11:31.17 | *** join/#asterisk voipbox (n=voipbox@193.170.41.114) |
| 11:31.55 | voipbox | nuh |
| 11:32.14 | voipbox | ist hier irgendjemand mit skills |
| 11:32.33 | xianlp | wtf someone calls himself voipbox |
| 11:33.53 | iDunno | maybe it's a box, with a voice in it, and it's someones IP? |
| 11:34.16 | *** join/#asterisk _reDruM (n=exiles@71-210-7-145.eugn.qwest.net) |
| 11:34.21 | XIN01OZ | lol |
| 11:34.26 | Ahrimanes | voipbox: doch.. aber sprechen sie englisch bitte |
| 11:34.47 | *** join/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it) |
| 11:34.52 | agx | good morning |
| 11:35.06 | XIN01OZ | iDunno eitheir |
| 11:35.52 | agx | i can call from remote 100@Asterisk_IP but i cannot register myself as client.. i get into the logs: Jan 3 13:35:14 NOTICE[26979]: chan_sip.c:7708 handle_request: Registration from '<sip:gallo-sjphone@172.16.1.4>' failed for ... any idea? it works when i'm onto the same net as the PBX |
| 11:36.17 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:37.05 | nfi|ermes | 39 => 2345,XXXXXXX YYYYYY,x.yyyyy@abc-de.it,,attach=no|saycid=yes|envelope=yes|delete=no|nextaftercmd=yes |
| 11:37.21 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:37.45 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 11:40.03 | zoa | royk, you are paranoid |
| 11:40.07 | RoyK | zoa: not at all |
| 11:40.18 | zoa | i've seen that message before |
| 11:40.18 | RoyK | zoa: I'm not coming through with the normal address |
| 11:40.33 | zoa | i see them come through from the normal address |
| 11:40.36 | RoyK | zoa: I sent a message over a day ago and it still hasn't come through |
| 11:40.44 | zoa | royk@karlsbakk.net |
| 11:40.47 | RoyK | zoa: then why are they not in the archive? |
| 11:40.48 | RoyK | not royk |
| 11:40.53 | RoyK | that's a secondary address |
| 11:40.58 | RoyK | roy@karlsbakk.net is the normal |
| 11:41.04 | zoa | thats the one you are using normally, no ? |
| 11:41.08 | RoyK | zoa: so, no, I am not paranoid |
| 11:41.17 | zoa | ive always seen royk |
| 11:41.20 | RoyK | roy@karlsbakk.net is the normal. royk@karlsbakk.net is a virtual |
| 11:41.27 | RoyK | zoa: I've never used that on the list |
| 11:41.31 | RoyK | before today |
| 11:41.41 | RoyK | and then I unregistered that from the list |
| 11:42.07 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:42.21 | RoyK | zoa: please beleive this. just look through the archives if you don't beleive me.... |
| 11:43.04 | nfi|ermes | anyone can help me to understand the reason why my voicemail say: incorrect password when it is not |
| 11:43.12 | tzafrir_laptop | RoyK, anyway, you can check if you're registered through the web interface |
| 11:43.24 | RoyK | tzafrir_laptop: I have |
| 11:43.28 | RoyK | tzafrir_laptop: Of course I have |
| 11:44.36 | RoyK | tzafrir_laptop: As I said: (a) I tried to send an email to the list from roy@karlsbakk.net, got a message from moderator. (b) registered with roy@briiz.no, sent message, got through, (c) tried to send a reply to the initial thread from yesterday from roy@briiz.no, got a message from moderator, registered with royk@karlsbakk.net, sent today's message, got through |
| 11:44.43 | RoyK | tzafrir_laptop: so I'm not paranoid |
| 11:44.47 | RoyK | tzafrir_laptop: not dreaming |
| 11:44.54 | RoyK | tzafrir_laptop: someone is censoring me |
| 11:45.08 | RoyK | fuck this |
| 11:46.00 | OloBola | I agree, it's all about you! |
| 11:48.10 | zoa | royk, nobody even reads those moderator messages |
| 11:48.22 | zoa | why would they moderate you and leave all the junk pass through ? :p |
| 11:50.35 | tzafrir_laptop | RoyK, mind if I quote one of your paragraphs above in a reply mail? (it contains all of the email addresses you mentioned) |
| 11:52.09 | OloBola | I tend to think it's all about me when it's really, really NOT about me. Keeps me warm at night. |
| 11:53.52 | XIN01OZ | voipbox do u speak english |
| 11:54.06 | voipbox | of course |
| 11:54.20 | RoyK | tzafrir_laptop: just do so |
| 11:54.24 | XIN01OZ | coo |
| 11:54.32 | RoyK | zoa: I don't know |
| 11:56.16 | XIN01OZ | tis good to have a voipbox around |
| 11:56.38 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:56.48 | XIN01OZ | check msgz |
| 11:59.42 | *** join/#asterisk pengyong (n=lala@218.93.102.82) |
| 12:00.43 | agx | Q: how do i can use my sip client when outside of the office? it say: failed registration into the logs; instead it works perfectly when inside the office LAN. |
| 12:00.43 | XIN01OZ | im german and I cant speak it not even prolly if I tried |
| 12:01.11 | zoa | woher geht der bus ? |
| 12:01.32 | voipbox | you are only 'half-german' so don't care :) |
| 12:02.21 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 12:03.02 | XIN01OZ | well my daughter is half-half so do i need know now? |
| 12:03.18 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:03.51 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 12:04.16 | XIN01OZ | :) |
| 12:04.16 | voipbox | hm, do you think to know it would be useful =? |
| 12:04.47 | XIN01OZ | very much so... we shall visit next year |
| 12:05.13 | voipbox | everybody will understand english |
| 12:05.22 | Ahrimanes | no habla inglese |
| 12:05.45 | *** join/#asterisk docelm0 (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 12:05.47 | voipbox | ? |
| 12:05.55 | docelm0 | Whadup whadup!?!?! |
| 12:06.02 | XIN01OZ | i did not expect that at all |
| 12:06.36 | XIN01OZ | kind of a bummer |
| 12:06.45 | docelm0 | ?? |
| 12:07.13 | XIN01OZ | everybody in Germany will understand english |
| 12:07.18 | voipbox | yes shouldn't, but you will have a good time also without it - it's complex, so it would take you quite some time so learn it |
| 12:07.31 | voipbox | to |
| 12:07.49 | XIN01OZ | all the better then |
| 12:08.17 | XIN01OZ | good to know thanks |
| 12:08.22 | voipbox | do you life in us ? |
| 12:08.30 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 12:08.39 | XIN01OZ | yep u over there currently? |
| 12:09.02 | voipbox | no, had been there a few month ago |
| 12:09.14 | XIN01OZ | ah yeah where now |
| 12:09.18 | voipbox | austria |
| 12:09.28 | docelm0 | Tampa Florida! YAY! |
| 12:09.35 | voipbox | would like it better to be florida again |
| 12:10.16 | XIN01OZ | mobile, al 1 hour away |
| 12:10.44 | voipbox | i wouldn't be that cold then :) |
| 12:11.07 | XIN01OZ | <--hasnt been in real cold yet |
| 12:11.15 | voipbox | great country, really miss the language and people |
| 12:11.38 | docelm0 | 70 farenheit right now. Gonna be a gorgeous day |
| 12:11.50 | voipbox | we have a few degress (celsius) minus now - enjoy the sun :) |
| 12:11.51 | agx | Q: is there a real #asterisk channell ? this seems a fake one... O.o |
| 12:12.04 | XIN01OZ | will be going to Taiwan soon |
| 12:12.05 | iDunno | this is a real channel. with real people. |
| 12:12.10 | docelm0 | agx go sit in the corner.. Whats your issue |
| 12:12.40 | agx | i've problem refistering a sip client outside the PBX LAN |
| 12:12.45 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 12:12.50 | XIN01OZ | looking forward to all the voip equipment i should be able to get my hands on |
| 12:12.51 | voipbox | for hollidays ? |
| 12:13.10 | Ahrimanes | refistering? |
| 12:13.37 | agx | registering* |
| 12:13.43 | dpryo | Somebody know of a sip-utility to initiate a connection to a user and then send an audiofile? (netcat for sip:) |
| 12:13.44 | Ahrimanes | ah |
| 12:13.55 | XIN01OZ | ah no setting up server |
| 12:14.20 | agx | it works if i'm inside the LAN; but when outiside it doesnt... server log say: registration failed (i use auth/passwd) |
| 12:14.30 | XIN01OZ | what holidays will be going on over there u know |
| 12:14.54 | docelm0 | agx, does it work inside the lan? |
| 12:15.04 | docelm0 | if it works inside and not out you probably have a nat issue.. |
| 12:15.04 | agx | yes |
| 12:15.20 | Ahrimanes | thazza: i think it's a suppository |
| 12:15.21 | agx | when outside i can call extension like 10@EXTERNAL_IP |
| 12:15.28 | agx | but registration fails |
| 12:15.59 | XIN01OZ | happy new year though.. will be first server .. only 1 line for friend |
| 12:16.29 | docelm0 | there shouldnt be an issue inside or out. I have a 100 station asterisk PBX at my office w/ 15 stations not in the local network and they work fine. |
| 12:16.57 | agx | what params you use for them inside sip.conf ? |
| 12:17.11 | docelm0 | I dont use sip.conf. I use realtime. |
| 12:17.37 | docelm0 | I only use sip.conf for static end points. |
| 12:17.47 | agx | what is realtime? ^^ |
| 12:18.07 | docelm0 | ~wiki |
| 12:18.21 | agx | ok |
| 12:18.26 | NewSole | jbot realtime |
| 12:18.29 | jbot | i heard realtime is http://www.voip-info.org/wiki-Asterisk+RealTime |
| 12:18.35 | docelm0 | ~wiki realtime |
| 12:18.52 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 12:18.55 | docelm0 | ahh they changed the bot on me.. |
| 12:18.59 | agx | i use this in sip.conf: [gallo-sjphone] context=interni type=friend qualify=30000 username=gallo-sjphone secret=sjpass regexten=201 host=dynamic |
| 12:19.23 | agx | maybe this work only when i'm inside the lan... and there is something i'm too noob to know :))) |
| 12:19.27 | docelm0 | dude if you actually get 30seconds of lag on a sip phone you might as well give up |
| 12:20.01 | docelm0 | How is your asterisk server config'd.. Do you have dual NIC's one with private and one with public IP? |
| 12:20.08 | docelm0 | brb.. need sugar for cappuccino |
| 12:20.38 | agx | LAN <---> PBX on DMZ <--> PIX-FW <--> I*NET <--> Notebook |
| 12:20.38 | Ahrimanes | mm black coffee |
| 12:20.58 | Ahrimanes | hm pix making some fun then+ |
| 12:21.09 | XIN01OZ | u could most likely drink brew made from betel nuts |
| 12:21.22 | RoyK | agx: pix setup to do stateful SIP? |
| 12:21.25 | RoyK | that's funny |
| 12:21.31 | RoyK | it can't handle double NAT iirc |
| 12:22.00 | Ahrimanes | yeah if pix is nat'ing external ip's to internal ip's on the dmz there'd be trouble |
| 12:22.05 | ManxPowe | The best thing to do is disable the SIP NAT translations on the PIX. |
| 12:22.13 | agx | dunno, i just asked them to redirect 5060 and 10000-12000 from extIP to PbxDmxIP |
| 12:22.24 | *** join/#asterisk kshumard (n=kshumard@gateway.digium.com) |
| 12:22.27 | agx | and it works both calling from inside or from outside |
| 12:22.30 | ruza | where can i set permissions for wav files created in voicemail box ? |
| 12:22.34 | Ahrimanes | ah nat it is.. |
| 12:22.39 | agx | i can call directly from sip client 100@myIP |
| 12:22.39 | ManxPowe | ruza, you don't. |
| 12:22.48 | ruza | ManxPowe: ?! |
| 12:23.21 | agx | the only problem i had is with the registration (not the calls) when i'm outside |
| 12:23.24 | ruza | ManxPowe: how can i set vmail.cgi to have permissions to delete msgs in voicemail than ? |
| 12:23.40 | ManxPowe | ruza, You have to run asterisk as the user vmail.cgi runs as. |
| 12:23.57 | docelm0 | agx you have biggier issue if you can work from inside and not out. What do you have setup for host in your sip.conf? |
| 12:24.07 | ManxPowe | ruza, It's prolly covered on the Wiki |
| 12:24.37 | docelm0 | Num Num.. Starbucks french vanilla Cappuccino.. |
| 12:24.40 | docelm0 | What a rush! |
| 12:25.05 | ruza | ManxPowe: where ? |
| 12:25.11 | agx | docelm0: i've set all the sip client to use host=dynamic + username and secret password |
| 12:25.48 | *** join/#asterisk guyee (n=izomtrik@nextra.nudli.equitas.hu) |
| 12:25.57 | agx | could it be an sjPhone bug maybe? i should test with a different program too? |
| 12:25.58 | docelm0 | Then you should be good to go inside and out.. |
| 12:26.14 | ruza | ManxPowe: maybe have apache user in asterisk group |
| 12:26.15 | docelm0 | if your double nat'd that would keep the call from working not registering |
| 12:26.27 | docelm0 | Apache? What bout it? |
| 12:26.45 | ruza | docelm0: vmail.cgi |
| 12:26.51 | agx | well thanks, i'll do some more tests maybe turning sip debug on |
| 12:27.03 | guyee | NE1 knows how to get caller ID work on outgoing connections with ooh323? |
| 12:27.04 | docelm0 | What do you wanna know? |
| 12:27.15 | agx | lunch time, bye and thanks you very much |
| 12:27.18 | docelm0 | Simplest way is to use make vmail something in the makefile |
| 12:27.24 | docelm0 | lunch??? |
| 12:27.30 | Ahrimanes | agx: well having nat on the dmz is a problem |
| 12:27.33 | *** part/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it) |
| 12:27.56 | ruza | docelm0: i want to be able delete msgs from voicemail via web |
| 12:28.06 | ruza | apache runs under user apache |
| 12:28.28 | iDunno | or www-data |
| 12:28.29 | ruza | permissions in /var/spool/asterisk are asterisk:asterisk |
| 12:28.29 | pr0m | ruza: me too. *stomp* *huff* |
| 12:28.32 | iDunno | (on debian) |
| 12:28.43 | ruza | iDunno: doesnt matter |
| 12:28.59 | ManxPowe | ruza, I'm sure the wiki has information on it. |
| 12:29.00 | ruza | iDunno: just different group |
| 12:29.05 | iDunno | erm - you could use cgiwrapd |
| 12:29.07 | ruza | ManxPowe: i cannot find it :) |
| 12:29.18 | iDunno | and write a cgi that then runs as the asterisk yser. |
| 12:29.21 | iDunno | erm - user. |
| 12:30.01 | Ahrimanes | the dark side you seek |
| 12:30.12 | ManxPowe | ruza, http://www.voip-info.org/wiki-Asterisk+administration |
| 12:30.32 | ruza | iDunno: how ? |
| 12:31.10 | ManxPowe | ruza, specifically http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root |
| 12:31.19 | iDunno | ruza: cgiwrapd is (similar) to suexec. |
| 12:32.55 | guyee | hey guyz, am I the only one using ooh323? :) |
| 12:33.36 | ManxPowe | guyee, many people these days use the H323 channel driver in asterisk-addons |
| 12:34.12 | *** join/#asterisk coppice (n=chatzill@212.201.17.210.dyn.pacific.net.hk) |
| 12:34.38 | jeffik | guyee: I am using h.323 |
| 12:35.04 | guyee | yep. asterisk-addons-1.2.1/asterisk-ooh323c. that's what I try to use. and it's almost perfect... almost |
| 12:35.20 | guyee | i cannot get caller id work on outgoing calls |
| 12:35.28 | ManxPowe | guyee, the guy that maintains it is pretty good about bug fixes and stuff. |
| 12:35.46 | ManxPowe | guyee, ask on the mailing list, see if anyone else has had a similar problem. |
| 12:36.40 | jalsot | hi |
| 12:36.48 | guyee | I'll have to... I just hoped that it's trivial enough to find the answer here :) |
| 12:37.32 | jalsot | is there a way to analyze IAX2 call streams? e.g. captured by ethereal [I mean, to see how many packets were lost, out of order, etc.] |
| 12:37.53 | ManxPowe | guyee, NOTHING with H323 is trivial |
| 12:38.19 | ManxPowe | jalsot, since IAX2 is UDP, the OS won't have that info. |
| 12:39.06 | guyee | ManxPowe: I have to agree :) |
| 12:39.10 | ManxPowe | jalsot, I dunno if ethereal can figure that out for IAX2 or not. I seem to recall an IAX2 specific plugin it, but I don't know where. |
| 12:40.38 | jalsot | ethereal can analyze IAX2 packets |
| 12:40.50 | jalsot | but not the whole stream, as it is available for RTP |
| 12:41.13 | jalsot | I meant, a tool, which can check timestamps and from those information make a summary |
| 12:41.38 | jalsot | I can make it by hand, however it is a crazy task for 40 concurrent calls and 20ms frames |
| 12:42.30 | jalsot | btw, we are experiencing problems on LAN with IAX2, so for this reason I would like to check if packets are arriving/going out right |
| 12:43.05 | jalsot | maybe somebody can help in that issue? |
| 12:43.49 | Ahrimanes | jalsot: http://lists.digium.com/pipermail/asterisk-dev/2003-November/002336.html ? |
| 12:43.55 | jalsot | show iax2 netstats shows ugly numbers as lost percentage when we go over about 30-40 concurrent calls (IAX2/g711a) on 100mbps LAN |
| 12:44.35 | jalsot | Ahrimanes: isn't it what is integrated into ethereal? |
| 12:44.43 | nfi|ermes | i have had congestion |
| 12:44.55 | nfi|ermes | no sip extensions can log in |
| 12:45.07 | Ahrimanes | jalsot: not sure.. seems an addon |
| 12:45.32 | jalsot | http://www.ethereal.com/docs/dfref/i/iax2.html |
| 12:46.00 | ManxPowe | jalsot, double check the duplex mode the interfaces are in. |
| 12:46.23 | *** join/#asterisk trixter (n=trixter@65.172.209.246) |
| 12:46.29 | ManxPowe | I had a problem where my Cisco router detected the wrong duplex on the port going to my DSL modem and something similar happend with only 2 calls. |
| 12:46.57 | jalsot | hmmm |
| 12:47.11 | ManxPowe | The problem showed up as collisions on s (supposed to be) full duplex interface. It's prolly not your problem, but it may be easier to check than trying to decode an IAX2 stream. |
| 12:47.29 | jalsot | the toplogy: Catalyst 3548 has about 9-10 SMC switches connected to it and the asterisk server |
| 12:47.46 | jalsot | operator stations with iax2 softphones are connected to those SMC switches |
| 12:47.48 | ManxPowe | jalsot, EEEEWWWWW! |
| 12:48.03 | jalsot | unfortunately that's at one of our customer |
| 12:48.12 | ManxPowe | jalsot, can you plug asterisk and most of the phones into the cat 3548? |
| 12:48.23 | ManxPowe | i.e. eliminte the SMC switches from the mix? |
| 12:48.28 | jalsot | I said EHHH for SMC as well, but I have to argue to customer :( |
| 12:48.37 | jalsot | unfortunately they cannot :( |
| 12:48.48 | pr0m | what's a catalyst 3548? |
| 12:48.52 | jalsot | Cisco |
| 12:48.58 | pr0m | ok |
| 12:49.16 | pr0m | i take it that it does load balance well? |
| 12:49.20 | newl | the customer is always right, even with they're wrong and when they realize they're wrong, you get extra money to fix. It's a win win situation. 8) |
| 12:49.27 | ManxPowe | jalsot, We are slowly getting rid of all our odd mixture of switches and moving to Cat 550x (cheap on eBay and will work for a long time. |
| 12:49.29 | jalsot | and of course, they say, the Call center is bad, because of choppy sound |
| 12:49.44 | *** join/#asterisk pengyong (n=lala@222.185.196.119) |
| 12:49.54 | ManxPowe | jalsot, Asterisk does not have a SIP jitterbuffer. |
| 12:50.04 | ManxPowe | Can you try turning on iax2 jitterbuffer? |
| 12:50.15 | jalsot | ManxPowe: so you don't have good experiences with SMC or SMC like switches? |
| 12:50.26 | jalsot | checking... |
| 12:51.03 | jalsot | right we have: |
| 12:51.04 | jalsot | jitterbuffer=yes |
| 12:51.04 | jalsot | forcejitterbuffer=no |
| 12:51.20 | ManxPowe | jalsot, I think SMC makes the Dell branded switches. Our vendor convinced our MIS manager to buy two 48-port Dell switches. Suddenly at random times people could not not connect to the Samba server. power cycle the switches and they could connect again. |
| 12:51.21 | jalsot | other settings are the defaults |
| 12:51.35 | ManxPowe | Replaced with Cisco, the problem didn't happen again. |
| 12:52.02 | coppice | ManxPower: lots of switches do that when the power hiccups |
| 12:52.29 | jalsot | ManxPowe: and what can be wrong with those switches? |
| 12:52.50 | zoa | i have the same stuff with cheap 3com switches |
| 12:53.02 | zoa | sometimes they just stop working for a while |
| 12:53.03 | jalsot | customer says: it must handle 80x60=4.8Mbps without any problems |
| 12:53.33 | *** join/#asterisk ErMeS|Work (n=ermsewrk@217.220.121.62) |
| 12:53.34 | coppice | yeah. *many* do this. I have a planet switch here behind a UPS, and it still does this |
| 12:53.35 | jalsot | they cannot handle small packets? or what can be wrong? |
| 12:53.44 | pr0m | load balancing. :) |
| 12:54.00 | ManxPowe | jalsot, no idea, they are now on the junk pile and the MIS manager no longer argues with me when I tell them to use Cisco |
| 12:54.05 | pr0m | sorry. i'm quite partial to openbsd's altq in pf. |
| 12:54.10 | jalsot | 50pps/call is about 3000pps for 60 calls, that doesn't seem to be so much |
| 12:54.31 | ManxPowe | jalsot, best to check the specs on the switches. |
| 12:54.36 | pr0m | unless they are coliding because they are not routed... they are switched. |
| 12:54.45 | ManxPowe | jalsot, are these calls going between the same two Asterisk servers? |
| 12:54.47 | pr0m | might need a router in the middle. |
| 12:54.47 | jalsot | I saw some thousands of pps in specs :( |
| 12:55.00 | pr0m | for real-time apps? |
| 12:55.14 | jalsot | no, iax softphones with one central asterisk |
| 12:55.31 | pr0m | big difference with real-time apps. not just a regular network load |
| 12:55.41 | jalsot | [v1.2 r7406] |
| 12:55.50 | pr0m | packets probably need to be prioritized. are you running a vlan? |
| 12:55.52 | ManxPowe | pr0m, I think he said they were getting lost packets, not just jitter |
| 12:56.01 | jalsot | so any idea how to argue to the customer? |
| 12:56.20 | ManxPowe | jalsot, does the SMC tell you anything? |
| 12:56.22 | pr0m | hmmm |
| 12:56.29 | ManxPowe | I would think it would log SOMETHING. |
| 12:56.48 | jalsot | unfortunately they cannot believe that those switches are not suitable for such usage and pointing with fingers to asterisk :( |
| 12:57.16 | ManxPowe | jalsot, the Asterisk server runs nothing else? |
| 12:57.20 | pr0m | manxpowe: if latency due to collisions increases the buffer beyond max... then packets are dropped in real-time apps. |
| 12:57.47 | ManxPowe | pr0m, *nod* I already told him to check his duplex settings |
| 12:57.51 | pr0m | manx: or if the packet arrives too late then it's dropped also. |
| 12:57.56 | jalsot | to be honest, they have usually SMC EZ6508TX, some D-Link D-Link DES-1016D and Ovislink FSH16T+ |
| 12:58.16 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:58.46 | jalsot | ManxPowe: it has monitoring and calls are going out on 2xTE110P |
| 12:58.47 | pr0m | what sort of netstats does the catalyst offer? |
| 12:59.01 | ManxPowe | jalsot, there MUST be a PC that's experiencing the problem that can plug into the same switch as Asterisk? |
| 12:59.12 | ManxPowe | jalsot, Monitoring would do it! |
| 12:59.21 | ManxPowe | are all calls being monitored? |
| 12:59.22 | jalsot | pr0m: unfortunately we don't have access to that switch, right waiting for some reports from them |
| 13:00.15 | ManxPowe | I wish the zaptel drivers could provide information on INTERRUPT jitter |
| 13:00.38 | jalsot | ManxPowe: we didn't find any relation with client machines and calls going bad - just the amount of calls |
| 13:00.42 | coppice | I wish they could provide any useful info |
| 13:00.52 | jalsot | ManxPowe: yes, usually all calls are monitored with monitor |
| 13:01.14 | ManxPowe | jalsot, have you tried turning off monitoring? |
| 13:01.23 | jalsot | recorded calls have all great quality from the zap side |
| 13:01.52 | pr0m | you could try snorting icmp packets on a bridged switch port to see if anything "obvious" is going on. |
| 13:02.01 | jalsot | ManxPowe: hmmm, I think not yet |
| 13:02.03 | *** join/#asterisk brockj49464 (n=brockj49@31.111.dhcp.hope.edu) |
| 13:02.05 | pr0m | wait. you said that you don't have "access" to the switch. |
| 13:02.25 | pr0m | hehehe. |
| 13:02.38 | ManxPowe | jalsot, SCSI or IDE or SATA for the interface for the hardd drive? |
| 13:02.45 | mutilator | ya know whats pathetic, 11-14% of our customers get lost carriers daily on dialup, we tried changing back to our portmasters because the lost carrier rate went down when we did that |
| 13:02.47 | pr0m | good thing you used a $2 bill. :-) |
| 13:02.59 | mutilator | but come to find out the portmasters are reporting lost carriers as user-request disconnects |
| 13:03.02 | jalsot | SATA drives with 3Ware 9500 RAID5 |
| 13:03.20 | mutilator | so our Total control boxes aren't junk like we thought |
| 13:03.21 | ManxPowe | jalsot, Egads man! You are not getting HDLC abort errors?????????????? |
| 13:04.06 | ManxPowe | jalsot, using onboard LAN or an addon card on the Asterisk server? |
| 13:04.10 | jalsot | ManxPowe: didn't experienced |
| 13:04.26 | pr0m | huh. irq errors? |
| 13:04.32 | jalsot | ManxPowe: it's a SuperMicro server with onboard Intel Etherexpress 1000 |
| 13:04.45 | jalsot | Intel Corp. 82546GB Gigabit Ethernet Controller |
| 13:05.01 | ManxPowe | jalsot, Try disableing the onboard lan and put in a PCI LAN card. |
| 13:05.07 | ErMeS|Work | AMP d3eleted my configuration files !!! |
| 13:05.14 | pr0m | could be.... what's top say about sysload on the asterisk server? are there alot of irq-bound cpu cycles? |
| 13:05.18 | ErMeS|Work | help me to recover them please !!! |
| 13:05.49 | jalsot | ManxPowe: hmmmm, could that help? why? |
| 13:05.54 | ManxPowe | ErMeS|Work, We can't. AMP has a specific method for making sure your whatever_custom.conf files are not deleted. Since AMP is written by a 3rd party, we don't really know a lot about it here. |
| 13:06.06 | pr0m | hehe. torment. |
| 13:06.13 | ManxPowe | jalsot, it has helped many people. |
| 13:07.13 | jalsot | I will try... |
| 13:07.17 | jalsot | any other idea? |
| 13:07.18 | ManxPowe | jalsot, Onboard devices frequently lock interrupts for a long time in order to improve performance. That can prevent other things from happening like servicing Zaptel interrupts |
| 13:07.40 | jalsot | ManxPowe: could using NAPI help? |
| 13:08.01 | hackeron | I'm trying to make all extensions available externally with this: 'exten => XXXX,3,Macro(stdexten,${EXTEN},20)' but I'm still hearing "invalid extension", what am I doing wrong? |
| 13:08.16 | ManxPowe | Graphics cards (onboard or addon) almost always do that. So do some onboard LAN and onboard IDE and onboard SATA |
| 13:08.20 | [TK]D-Fender | jalsot : I do believe you are using the dreaded E1000 driver for that onboard NIC. Kill it if you know whats good for you.... |
| 13:08.36 | jalsot | what do you think, what is more probable? SMC switches sucks vs. monitoring sucks vs. onboard NIC sucks? |
| 13:08.56 | ManxPowe | jalsot, I have no idea. Try the easiest things to change first. |
| 13:08.58 | [TK]D-Fender | jalsot : The NIC is good, just that Digiums cards have a hissy-fit over them |
| 13:09.08 | *** join/#asterisk saftsack (n=oliver@p54A7DA9D.dip.t-dialin.net) |
| 13:09.26 | ManxPowe | [TK]D-Fender, He's NOT getting HDLC issues, he's getting poor audio quality at about 30-40 calls |
| 13:09.41 | coppice | the NIC is good, but its driver sucks |
| 13:09.46 | [TK]D-Fender | hackeron : you forgot the "_" before your "XXXX". Its required to indicate that its a pattern. |
| 13:10.06 | newl | all clone hardware sucks..always has, always will until they work out true Autoconfig[tm] and stop trying to have IRQ wars. :) |
| 13:10.08 | ManxPowe | I found another one for the Top Ten Newbie Questions/Issues! |
| 13:10.23 | jalsot | coppice: is there a better driver? I checked intel side and there is bit newer driver |
| 13:10.27 | [TK]D-Fender | coppice : I've never had a problem with it or anything in combination except Digium cards... Now ask me where I'd lay blame.... |
| 13:10.46 | pr0m | but without more info it seems like a crap shoot. |
| 13:10.51 | ManxPowe | jalsot, you now have MANY things to check. |
| 13:10.56 | pr0m | heheh. |
| 13:11.01 | coppice | newl: that isn't the problem. the driver hangs on to the interrupt and twiddles its thumbs blocking everything else |
| 13:11.25 | pr0m | more info! |
| 13:11.37 | pr0m | hehe. if all else fails look at the computer. |
| 13:11.40 | newl | coppice: that'd be a poorly written driver then indeed. |
| 13:11.40 | hackeron | [TK]D-Fender: hmm, now it gives up after the first 2 digits |
| 13:11.44 | pr0m | ;-) |
| 13:11.48 | coppice | jalsot: if they have a newer driver they may have improved this. |
| 13:12.01 | ManxPowe | jalsot, just for fun, run a ping from a PC with an IAX2 phone to the asterisk server (ping -t) and watch the jitter/latency/etc. Remember many switches and routers give ICMP a low priority, but it might be interesting to see. |
| 13:12.08 | [TK]D-Fender | hackeron : must be something else in your dial-plan |
| 13:12.16 | pr0m | manxpowe: smart. |
| 13:12.21 | jalsot | coppice: ok, I will check new driver - just from changelog didn't think it could help, but will try |
| 13:12.22 | ManxPowe | hackeron, what devices are the calls coming in on?> |
| 13:12.27 | pr0m | thpppppp. |
| 13:12.32 | [TK]D-Fender | hackeron : Pastebin the whole thing and I'll take a look. |
| 13:12.33 | [TK]D-Fender | ~pb |
| 13:12.36 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:12.37 | coppice | in the zapata 1 days we had trouble with a 3-COM LAN card that did this. with most other LAN cards things ran cleanly. now intel are causing similar problems |
| 13:13.06 | jalsot | what NIC should I buy than? |
| 13:13.14 | pr0m | don't spend money yet. |
| 13:13.21 | pr0m | need more info! blah! |
| 13:13.27 | hackeron | ManxPowe: [TK]D-Fender: http://pastebin.com/488565 |
| 13:13.35 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 13:13.36 | jalsot | pr0m: what info do you need? :) |
| 13:13.54 | pr0m | try manxpowe's suggestion. |
| 13:14.02 | jalsot | which one :) |
| 13:14.03 | pr0m | get some actual number to work with. |
| 13:14.46 | pr0m | ping with equivalent voip packet sizes on the network from one of the softphone stations to the asterisk server. |
| 13:14.59 | hackeron | ManxPowe: [TK]D-Fender: basically, I dial the incoming number, the system waits for extension, I give it the extension, I want the stdexten macro to run for the given extension |
| 13:15.10 | pr0m | check irq sysload on the asterisk server to see if the nic is binding up the cpu. |
| 13:15.10 | [TK]D-Fender | hackeron :[stdexten]; should be written [macro-stdexten] |
| 13:15.15 | jalsot | conclusion: 1. Cisco switches in place of low-end SMC/D-link, 2. monitor off, 3. PCI NIC in place of onboad E1000, 4. NIC kernel driver upgrade |
| 13:15.30 | [TK]D-Fender | hackeron : poorly defined macro kills the calling line... |
| 13:15.51 | [TK]D-Fender | OH, and # |
| 13:15.52 | [TK]D-Fender | exten => _XXXX,3,Macro(stdexten,${EXTEN},20) |
| 13:15.52 | [TK]D-Fender | # |
| 13:15.58 | pr0m | i'd try the least expensive of those options first. |
| 13:15.59 | jalsot | pr0m: 17: 1470270398 34680848 IO-APIC-level eth0 |
| 13:16.06 | [TK]D-Fender | TRhat should be priority "1", not "3" |
| 13:16.30 | jalsot | that's a Dual Xeon 2.8GHz |
| 13:16.59 | pr0m | vmstat |
| 13:17.00 | [TK]D-Fender | hackeron : While you're at it you are letting people calling you dial OUT... *NOT HEALTHY* |
| 13:17.27 | hackeron | [TK]D-Fender: hmm, how do I stop that? |
| 13:17.28 | pr0m | hehe. i did that when i first setup my extensions. |
| 13:17.48 | pr0m | basically callthrough without DISA. ;-P |
| 13:18.35 | hackeron | [TK]D-Fender: and what do you mean by #? |
| 13:18.41 | [TK]D-Fender | hackeron : move that into another context. make 1 context for your internal extensions (numbering them specifically), one for your outgoing, one for your phones (which INCLUDE internal + external), and one for your INBOUND calls (which will INCLUDE internal only) |
| 13:18.58 | [TK]D-Fender | hackeron : The "#"'s were just cut and pasted from your pastebin, sorry. |
| 13:19.34 | jalsot | pr0m: procs -----------memory---------- ---swap-- -----io---- --system-- ----cpu---- |
| 13:19.37 | jalsot | <PROTECTED> |
| 13:19.37 | jalsot | <PROTECTED> |
| 13:20.04 | pr0m | gahhhhhhhhhhh |
| 13:20.17 | pr0m | waiting!? hello? |
| 13:20.27 | jalsot | that's all |
| 13:20.40 | jalsot | right there is no load on the server |
| 13:20.44 | pr0m | no. i mean... "waiting" in cpu load. |
| 13:20.53 | jalsot | customer stopped all campaigns.. |
| 13:21.09 | pr0m | oh. oops. |
| 13:21.11 | jalsot | you mean uptime? |
| 13:21.17 | hackeron | [TK]D-Fender: also, I want all internal extensions available externally (except being able to dial out), any way to do that without naming each extension specifically? |
| 13:21.18 | pr0m | hehe. i had the columns wrong. sorry. |
| 13:21.26 | *** join/#asterisk Lloydie-t (n=lloydie-@thomasclan.plus.com) |
| 13:21.28 | pr0m | looks good. to me. :) |
| 13:21.42 | [TK]D-Fender | hackeron : use INCLUDE's like I mentioned. |
| 13:21.56 | [TK]D-Fender | hackeron, list me your VALID extension #'s (phones) |
| 13:22.08 | pr0m | not likely an irq problem if 'waiting' doesn't get saturated. |
| 13:22.17 | Lloydie-t | Hi All |
| 13:22.19 | [TK]D-Fender | (XXXX) Allows bad #'s to be dialed |
| 13:22.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:22.27 | jalsot | pr0m: right there are no calls in that box |
| 13:23.00 | pr0m | ok. not very good example of average load then? |
| 13:23.12 | Lloydie-t | I am having problems with mysql realtime. I get the following error when I try to register a client |
| 13:23.16 | Lloydie-t | config.c:920 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 13:23.26 | hackeron | [TK]D-Fender: at the moment there are 2, there will be 47 -- so far its 6272 and 7662. |
| 13:23.27 | jalsot | average load was always under 5.00 |
| 13:23.37 | Lloydie-t | Any Ideas? |
| 13:23.49 | jalsot | I read somewhere that on dual CPU system untill 10 it can be ok |
| 13:24.13 | jalsot | ManxPowe: thanks for help and ideas! |
| 13:24.16 | [TK]D-Fender | hackeron : Any special reason for the wide range? Matching 4 digits on a DID? |
| 13:24.18 | pr0m | under 5.00? 4.5 is kind stressed if you ask me. |
| 13:24.33 | hackeron | [TK]D-Fender: no, spells the person's name |
| 13:24.36 | jalsot | pr0m: yes, but it is just sometimes |
| 13:24.42 | newl | If that's the case, my box is sweating bullets at 11.45 then. :D |
| 13:24.47 | jalsot | usually because lame [with nice 19] |
| 13:24.48 | hackeron | [TK]D-Fender: (or the first 4 characters of their name) |
| 13:24.57 | coppice | jalsot: OK for what? telephony need first rate responsiveness |
| 13:24.57 | pr0m | well. maybe. i have a dual amd board... if i got the load over 4 then my terminal session started to lag. |
| 13:25.31 | jalsot | coppice: I read that on voip-info.org |
| 13:25.42 | coppice | and if a terminal session lags, voice has long since gone down the toilet |
| 13:25.50 | pr0m | sheesh. 11.45. |
| 13:25.55 | jalsot | :) |
| 13:26.10 | jalsot | I guess, nice 19 does not help than |
| 13:26.13 | coppice | jalsot: oh sorry. my mistake. you went straight to the authoratative source :-) |
| 13:26.29 | jalsot | so I should eliminate LAME than |
| 13:26.48 | jalsot | [what was my plan, but crazy customer wants mp3 :( ] |
| 13:26.54 | [TK]D-Fender | hackeron : gimme a sec, fixing it up for you. |
| 13:27.08 | lesouvage | I changed to isdn today. Should I connect a powersupply to the nt1 box to get a functioning isdn hfc connection or should there be another reason that the connection isn't working. I paste some relevant info to http://pastebin.ca/35611 . I'm without phone atthis moment so any hint is more than welcome. |
| 13:27.19 | hackeron | [TK]D-Fender: thanks for that, appreciate it! |
| 13:27.36 | jalsot | coppice: do you think, mixmonitor with sox to GSM [wav49] would give similar voice quality than mp3? |
| 13:27.47 | pr0m | brb. postoffice bound. |
| 13:28.05 | *** join/#asterisk tini (n=tini@193.170.41.114) |
| 13:28.18 | tini | I have question concerning the voicemail service |
| 13:28.42 | RoyK | tini: ask the question, and someone might give you an answer..... |
| 13:28.52 | tini | hope so - thx royk |
| 13:30.52 | tini | I configured the voicemail that it's possible to use it without password (otherwise the login is incorrect), but whatever I don't care about that - the problem is that I can real the VoiceMailMan but it doesn't react on my input, is there any configuration necessary ? |
| 13:31.39 | tini | hm, maybe it's my cisco phone, a 7960 ? |
| 13:31.55 | *** join/#asterisk eKo1 (n=bernd@63.245.57.70) |
| 13:32.03 | saftsack | The present kernel configuration has modules disabled. |
| 13:32.11 | saftsack | misdn says this to me but its not true |
| 13:32.13 | saftsack | howto fix it? |
| 13:32.17 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 13:32.36 | *** join/#asterisk jnandreae (n=jnandrea@d019010.adsl.hansenet.de) |
| 13:32.40 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 13:33.17 | hackeron | [TK]D-Fender: variable extension length will be great if possible btw :) |
| 13:33.42 | *** join/#asterisk caryon (n=caryon@p54A3B714.dip0.t-ipconnect.de) |
| 13:35.12 | saftsack | fixed it. i created a configfile |
| 13:35.57 | [TK]D-Fender | hackeron : Merry Christmas -> http://pastebin.com/488593 |
| 13:36.03 | *** join/#asterisk radar (n=alex@fire1.4synergy.com) |
| 13:36.49 | hackeron | [TK]D-Fender: heh, what a great present, exactly what I wanted :) |
| 13:36.55 | radar | meetme.conf says conf=>1234, but the asterisk cli means: *CLI> meetme lock 1234 : Jan 3 13:34:41 NOTICE[22312]: app_meetme.c:2100 admin_exec: Conference Number not found |
| 13:37.00 | [TK]D-Fender | hackeron : set your teliax accounts context to [incoming], and your phones to [myphones] |
| 13:37.35 | jalsot | pr0m: thanks for suggestions |
| 13:37.49 | jalsot | coppice: thanks for ideas |
| 13:38.26 | saftsack | i hate suse |
| 13:38.34 | saftsack | always 2.4 kernels here :( |
| 13:39.01 | hackeron | [TK]D-Fender: hmm, I thought autofallthrough=yes was the highly recommended option? |
| 13:39.02 | ManxPowe | > There is already a very good database for binary files, |
| 13:39.02 | ManxPowe | > > called "a filesystem" |
| 13:39.02 | ManxPowe | Is there any how-to for filesystem and Asterisk voicemail storage? |
| 13:39.08 | [TK]D-Fender | saftsack : I run 2.4 on all my Slackware systems and everything runs absolutely perfectly. |
| 13:39.17 | saftsack | do you have misdn? |
| 13:39.40 | [TK]D-Fender | hackeron : Not IMO...... |
| 13:39.43 | radar | anybody using MeetMe here? |
| 13:39.53 | ManxPowe | radar, yes |
| 13:40.00 | [TK]D-Fender | saftsack : if that makes a difference (which I'm beginning to suspect), then no. |
| 13:40.08 | radar | ManxPowe: can you help me? see question above |
| 13:40.17 | saftsack | i think that misdn just works on 2.6 |
| 13:40.39 | ManxPowe | radar, no. I don't static conference numbers |
| 13:41.00 | hackeron | [TK]D-Fender: hmm, any particular reason? |
| 13:41.06 | radar | ManxPowe: you make dynamic channels? |
| 13:41.14 | ManxPowe | exten => 3302,1,MeetMe(,pdq) |
| 13:41.32 | ManxPowe | That is all I have in addition to a /etc/asterisk/meetme.conf that contains only 1 line [rooms] |
| 13:41.40 | radar | mmmhm |
| 13:41.56 | radar | do you create them directly in the asterisk cli? |
| 13:42.13 | [TK]D-Fender | hackeron : I like controlling what happens on timeout & invalid extensions. LIke being able to put a counter for # of failed attempts to prevent people war-dialing extensions, or from sitting around and wasteing money on 1-800 calls at our expense <- |
| 13:42.33 | ManxPowe | radar, If you do a "show application meetme" in the Asterisk CLI and read up on the p, d, and q options you'll understand exactly what I'm doing. |
| 13:42.57 | hackeron | [TK]D-Fender: ah, makes sense |
| 13:43.03 | tini | hm |
| 13:43.14 | radar | ManxPowe: I can't find an example |
| 13:43.18 | [TK]D-Fender | hackeron : AEL & autofallthrough, were someones "nice idea" that I have never understood the need for. I see no added value for their existance. |
| 13:43.37 | ManxPowe | radar, did you do a "show application meetme" in the Asterisk CLI and read it? |
| 13:43.45 | *** join/#asterisk kart_179 (n=kart@200-181-212-144.mganm7003.dsl.brasiltelecom.net.br) |
| 13:43.54 | radar | ManxPowe: sure |
| 13:43.56 | ManxPowe | [TK]D-Fender, I like the idea of AEL, but it seems to be pretty buggy. |
| 13:44.03 | kart_179 | Who knows one softfone freeware or trial, that supports a g723 codec ? |
| 13:44.17 | ManxPowe | And you can load res_perl or res_js and pretty much do the same thing. |
| 13:44.27 | ManxPowe | kart_179, none because G723.1 is patented. |
| 13:44.42 | kart_179 | Oh cheat !!! |
| 13:44.55 | saftsack | [TK]D-Fender, make[2]: *** No rule to make target `modules'. Stop. |
| 13:45.03 | saftsack | is this a kernel 2.4 issue? |
| 13:45.04 | radar | ManxPowe: create a channel 1234: *CLI> meetme -d 1234 -pq |
| 13:45.15 | ManxPowe | Asterisk only supports G723.1 in "passthru" mode, which is pretty useless for most people. |
| 13:45.25 | ManxPowe | radar, What the hell are you doing? |
| 13:45.34 | ManxPowe | exten => 3302,1,MeetMe(,pdq) |
| 13:45.36 | ManxPowe | in extensions.conf |
| 13:45.39 | ManxPowe | that's all you need. |
| 13:46.03 | coppice | why does anyone want G.723.1 these days? |
| 13:46.27 | zoa | it sucks |
| 13:46.42 | ManxPowe | coppice, As far as I can tell the only reason is ignorance |
| 13:46.44 | zoa | kart_179, xpro used to have it for a while |
| 13:46.45 | [TK]D-Fender | saftsack : Beyond my experience, sorry :( |
| 13:47.03 | saftsack | didnt get you sry |
| 13:47.11 | coppice | its strange how many people fail to notice it sucks |
| 13:47.13 | ManxPowe | then dial 3302 to be connected to meetme and be prompted for a conference number to create |
| 13:47.24 | ManxPowe | coppice, Actually I don't think it sucks at all. |
| 13:47.33 | [TK]D-Fender | hackeron : So, everything working with the new extensions.conf? |
| 13:47.38 | coppice | then you have failed to notice too |
| 13:48.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:48.48 | iCEBrkr | ugh, do I have to get back to work today??? |
| 13:48.55 | radar | ManxPowe: oh, ok, got it. but calling meetme still says No active MeetMe conferences. do I have to put a user into a conference, first? |
| 13:49.05 | ManxPowe | But it didn't sound better or worse in any significant way to my untrained ear than G726, G729, or SpeeX |
| 13:49.11 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:49.13 | ManxPowe | radar, no. |
| 13:49.30 | coppice | ManxPower: there. you did fail to notice its serious problem |
| 13:49.32 | ManxPowe | radar, if you are getting that message then you are not running meetme with the options I gave you. |
| 13:49.40 | [TK]D-Fender | iCEBrkr : maybe those morons have finished the job and its a bright and sunny day waiting for you! |
| 13:49.43 | ManxPowe | coppice, What's SO bad about it? |
| 13:50.00 | pr0m | jalsot: welcome. :) |
| 13:50.03 | ManxPowe | Isn't G723.1 what AT&T tried to market as "TrueVoice". |
| 13:50.21 | iCEBrkr | [TK]D-Fender: I have a feeling they had those T1's turned up, but they were in alarm cuz I didn't have'm plugged in or the card configured correctly, so they're probably in loopback |
| 13:50.27 | coppice | ManxPower: its the DSP Group that market it as TrueVoice |
| 13:50.58 | [TK]D-Fender | iCEBrkr : this auto-loopback is utter BS so they don't have to stare at a red light all the time! Petty illusion of good service! |
| 13:50.59 | ManxPowe | coppice, It's either a different TrueVoice or AT&T licensed it. |
| 13:51.15 | coppice | ManxPower: it sounds nice. that isn't the problem. it makes everyone sound the same. that is the problem. identifying the speaker is very very hard |
| 13:51.24 | iCEBrkr | [TK]D-Fender: According to a few friends, a lot of telco's do that. |
| 13:51.39 | coppice | it must be the same truevoice, as its a registered trade mark |
| 13:51.41 | ManxPowe | coppice, Ah. So it sucks for weenies that have a fetish for conference calls and/or speaker phones? |
| 13:51.49 | radar | ManxPowe: I put that into /etc/asterisk/extensions.conf: exten => 3308,1,MeetMe(,pdq); and restarted asterisk. I also removed the old meetme.conf file, so that they don't confusing each other. |
| 13:52.15 | coppice | it sucks when you pick up a call from your lover, and have to ask who it is. its a terrible codec |
| 13:52.18 | ManxPowe | coppice, AT&T tried to counter Sprint's "pin drop" marketing stuff with a TrueVoice marketing stuff |
| 13:52.21 | [TK]D-Fender | iCEBrkr : I somehow believe that. |
| 13:52.32 | ManxPowe | radar, and then you dial 3308? |
| 13:52.45 | coppice | its main early use was for video conferencing, and it gets *really* confusing there |
| 13:53.04 | ManxPowe | Bah! Move someplace with Caller*ID 8-) |
| 13:53.11 | coppice | what's pin drop marketing? |
| 13:53.17 | ManxPowe | Unlike GSM, of course. |
| 13:53.19 | radar | ManxPowe: yes, but nothing happens... |
| 13:53.24 | hackeron | [TK]D-Fender: Hey, sorry, just cleaning some other files, about to test now |
| 13:53.29 | iCEBrkr | coppice: Don't you remember Sprints commercials? |
| 13:53.30 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:53.38 | ManxPowe | coppice, Sprint: We have a %100 fiber optic network! Calls so clear you can hear a pin drop!" |
| 13:53.40 | coppice | i'm not american |
| 13:53.42 | iCEBrkr | coppice: They provided so clear, you could hear a pin drop |
| 13:53.49 | iCEBrkr | oh |
| 13:53.50 | ManxPowe | iCEBrkr, coppice lives in HK |
| 13:53.53 | iCEBrkr | oops |
| 13:53.56 | radar | ManxPowe: I don't even get a message in CLI |
| 13:54.11 | coppice | Sprint was the only digital network at that time with low quality |
| 13:54.22 | puzzled | morning |
| 13:54.44 | hackeron | [TK]D-Fender: hmm, many errors: http://pastebin.com/488625 |
| 13:55.31 | ManxPowe | rad you should get something like this: |
| 13:55.34 | ManxPowe | <PROTECTED> |
| 13:55.34 | ManxPowe | <PROTECTED> |
| 13:56.10 | radar | hm damn |
| 13:56.34 | ManxPowe | the Zap/1-1 will be different, depending on what you are using to place the call |
| 13:57.11 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 13:57.17 | [TK]D-Fender | hac : oops. its unchanged from the pastebin still? |
| 13:57.42 | iCEBrkr | Well, I guess I'll do the mindless job of building/installing this server. |
| 13:57.47 | iCEBrkr | I really don't feel like code today. |
| 13:58.01 | [TK]D-Fender | hackeron : Oh, and what Tech are your phones? |
| 13:58.23 | iCEBrkr | ...and then maybe I can reconfigure this Sangoma card and get the T1s up and running today |
| 13:58.31 | hackeron | [TK]D-Fender: SIP, added it already :) -- but I dont have the recorded welcome messages or anything. Any defaults I can use? |
| 13:58.42 | iCEBrkr | Hrrrm, too ambitious sounding. |
| 13:58.58 | radar | ManxPowe: oh, now my client says 404, file not found (x-lite). :-) |
| 13:59.18 | hackeron | [TK]D-Fender: also, its outgoing, not outbound in the myphones include :) |
| 13:59.47 | ManxPowe | I GUESS I should brave traffic and head to a client site. |
| 13:59.53 | *** join/#asterisk brimston1 (n=brimston@pcp01534724pcs.huntsv01.al.comcast.net) |
| 14:00.31 | [TK]D-Fender | hackeron : here, add this whole context on the end and include it in [myphones} - http://pastebin.com/488634 |
| 14:00.41 | [TK]D-Fender | hackeron ;: DETAILS! |
| 14:00.43 | [TK]D-Fender | :p |
| 14:01.08 | [TK]D-Fender | hackeron : In there is a little recording macro so you can make your own ones. |
| 14:01.34 | *** join/#asterisk brockj49464_ (n=brockj49@22.105.dhcp.hope.edu) |
| 14:02.05 | hackeron | [TK]D-Fender: hmm: Jan 3 04:02:09 WARNING[8533]: pbx.c:4764 ast_add_extension2: Unable to register extension 's-BUSY', priority 2 in 'macro-stdexten', already in us |
| 14:03.04 | ManxPowe | hackeron, that's a pretty obvious message. |
| 14:03.25 | radar | ManxPowe: any other idea? |
| 14:03.28 | ManxPowe | points you to EXACTLY what the problem is, and even where to look. |
| 14:03.38 | ManxPowe | radar, I'm sorry, I do not have time to teach you Asterisk. |
| 14:04.02 | ManxPowe | you have two exten => s-BUSY,2 |
| 14:04.11 | ManxPowe | that was for hackeron |
| 14:04.21 | [TK]D-Fender | hack, maybe another oops... |
| 14:04.51 | [TK]D-Fender | hackeron : they should be 1,2,3, in order in that macro... forgot to fix that from a sample I gave someone else... |
| 14:05.03 | [TK]D-Fender | Check busy & unavail |
| 14:05.19 | [TK]D-Fender | renumber that and you'll be OK. |
| 14:05.54 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:06.07 | Cresl1n | ooh... haven't been in here for a while |
| 14:08.39 | [TK]D-Fender | hackeron : And you added the SIP/ in the line that CALLS the macro STDEXTEN, right? |
| 14:09.02 | [TK]D-Fender | hackeron : (if you do it there you can use the macro for multiple tech's) |
| 14:09.13 | *** part/#asterisk radar (n=alex@fire1.4synergy.com) |
| 14:09.21 | hackeron | [TK]D-Fender: yes, just plugging in a phone here in the UK to test, other phone in the US and no one there yet :) |
| 14:09.27 | *** join/#asterisk __Soul__ (n=Soul@87-196-13-46.net.novis.pt) |
| 14:09.33 | hackeron | [TK]D-Fender: multiple tech's? |
| 14:09.58 | [TK]D-Fender | hackeron : Ok, fire away. and use the *40 section to make your prompts (including the one it uses itself ) and move them into the proper folder |
| 14:10.14 | [TK]D-Fender | hackeron : in case you add IAX2 or ZAP extensions to your PBX |
| 14:15.13 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 14:16.08 | saftsack | comparison between signed and unsigned |
| 14:16.15 | saftsack | im getting thsi warning all the time |
| 14:16.18 | saftsack | what does it mean? |
| 14:16.37 | Ahrimanes | different types og integers |
| 14:17.02 | saftsack | that means? because here on my kernel compilation i get tons of this errors |
| 14:22.00 | hackeron | [TK]D-Fender: hmm, I dial *40, say something, hit # and nothing happens.. the connection is active and I can see Executing Record("SIP/7662-2117", "/tmp/asterisk-recording:ulaw") in new stack but how do I terminate recording? |
| 14:22.21 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:22.58 | [TK]D-Fender | "#" |
| 14:23.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:23.19 | santoshr | <PROTECTED> |
| 14:23.34 | santoshr | using sip and asterisk 1.2 |
| 14:24.00 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 14:24.28 | [TK]D-Fender | santoshr : as in dial directly in extensions.conf? |
| 14:24.37 | santoshr | yes |
| 14:24.54 | [TK]D-Fender | IIRC you aren't supposed to call an agent directly like that. |
| 14:25.20 | [TK]D-Fender | its only for use by Queue's and gets redirected to the context used in your queue def which could point ANYWHERE |
| 14:25.48 | santoshr | ok.. |
| 14:26.12 | [TK]D-Fender | Why would you ever want to call an agent directly throug that? |
| 14:27.46 | santoshr | then how do i open the channel |
| 14:28.17 | Whisk | <PROTECTED> |
| 14:28.29 | Whisk | not tried it post 1.2 but i should work i think |
| 14:28.30 | [TK]D-Fender | santoshr : What do you mean open the channel? Those phones should have noral Dial lines leading to their tech/name. |
| 14:28.39 | Whisk | make sure you've got the agent defined/logged in properly |
| 14:28.44 | [TK]D-Fender | The question again is WHY? |
| 14:29.14 | Whisk | it's useful if you have e.g. people using agentcallbacklogin and logging in on different phones |
| 14:29.22 | [TK]D-Fender | Typically any agent you have should have another NORMAL way of dialing their exte. |
| 14:29.25 | Whisk | you can tie a ddi into their agent login |
| 14:29.49 | [TK]D-Fender | Whisk : but he's talking about using it DIRECTLY. Not as being called by the app_queue |
| 14:29.49 | santoshr | Whisk: we are using agentcallbacklogin.. |
| 14:30.03 | Whisk | i know [TK]D-Fender |
| 14:30.08 | [TK]D-Fender | Whisk : ok, thats valid. Wierd, but valid. |
| 14:30.12 | Whisk | heh :) |
| 14:30.17 | santoshr | so.. |
| 14:30.23 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 14:30.30 | santoshr | wht woudl app_dial not dial |
| 14:30.39 | [TK]D-Fender | santoshr : So... can't help you there :) sorry.... |
| 14:30.48 | Whisk | check that your agents are logged in then and that you're specifying the right agent details etc |
| 14:31.02 | santoshr | yeah Whisk double checked tht.. |
| 14:31.58 | Whisk | and you are dialling AGENT/agentid? |
| 14:32.33 | santoshr | yes |
| 14:33.22 | santoshr | if dial the extension on which agent logged on.. it goes thru.. but the agent id .. it wont.. |
| 14:33.28 | Whisk | and if you do show agents they're logged on etc? |
| 14:35.08 | santoshr | yeah it does say tht the agent is avaialabe |
| 14:36.20 | santoshr | Whisk: ......... |
| 14:36.26 | *** join/#asterisk MnxPower (i=ewieling@1.sub-70-197-82.myvzw.com) |
| 14:37.55 | *** join/#asterisk umay (n=chris@65-37-2-236.nrp2.roc.ny.frontiernet.net) |
| 14:38.58 | [TK]D-Fender | hackeron : So, up and running? |
| 14:39.35 | Whisk | santoshr - i've just tried it on a post 1.2 install and it works fine |
| 14:39.38 | hackeron | [TK]D-Fender: not yet, something up with phone, wont let me change user login/password.. Its a grandstream GXP-2000 - such a pos |
| 14:39.40 | Whisk | i'd check for something silly |
| 14:40.20 | [TK]D-Fender | hackeron : Grandsuck strikes again! |
| 14:40.55 | hackeron | [TK]D-Fender: not very good phones, eh? :) |
| 14:41.16 | santoshr | agents.conf has only one agent as o fnow.. supposedly 12 and in extensions.conf thr a exten to call agenttcallbacklogin ---> it logs in when i try to do so |
| 14:42.21 | saftsack | hugo-v6, hi |
| 14:44.22 | Whisk | what's your dial line look like |
| 14:44.28 | santoshr | i get this when i am dialing <<<<<<<<<<<Dial("Local/12@default-e07f,2", "Agent/12") >>>>>>>> |
| 14:45.00 | santoshr | exten => 11,n,AgentCallbackLogin |
| 14:45.01 | santoshr | exten => 12,1,Dial(Agent/12) |
| 14:46.39 | santoshr | and in agents.conf agent => 12,12,x |
| 14:46.52 | *** join/#asterisk Splas (i=jwb@206.252.198.100) |
| 14:47.20 | Whisk | you're not logging in as 12 are you? |
| 14:47.29 | santoshr | yes ? |
| 14:47.41 | Whisk | that's gonna cause some wierd loop then |
| 14:48.05 | Whisk | cos the agent channel will call 12 which calls the agent channel etc |
| 14:48.38 | hackeron | [TK]D-Fender: hmm, I see "IAX2/teliax-2 stopped sounds" when I dial a number and I cant hear the person on the other end, I also see Spawn extension (myphones, 12127867577, 1) exited non-zero on 'SIP/7662-542f' -- any ideas? |
| 14:49.12 | zoa | hey santoshr: you are the same who posted this to the asteriskguru forum ? |
| 14:49.56 | santoshr | ya a collegue .. of mine.. |
| 14:50.02 | zoa | aha |
| 14:50.12 | zoa | when we have some time i'll have a go |
| 14:50.23 | santoshr | ohhh common.. |
| 14:51.06 | [TK]D-Fender | hackeron : Pastebin the whole call |
| 14:51.59 | hackeron | [TK]D-Fender: http://pastebin.com/488688 |
| 14:52.17 | hackeron | [TK]D-Fender: wait, sorry, missed the last line |
| 14:52.37 | hackeron | [TK]D-Fender: http://pastebin.com/488690 |
| 14:53.13 | [TK]D-Fender | hackeron : No menu at all! |
| 14:53.23 | hackeron | [TK]D-Fender: no, I'm ringing from phone to someone |
| 14:53.25 | Whisk | santoshr - stick your extensions.conf /agents.conf and the cli for the call in pastebin and i'll have a look - it should be pretty simple to get it working |
| 14:53.41 | santoshr | ok wait.. |
| 14:53.56 | [TK]D-Fender | hackeron : Don't think we configured that phone # in [incoming], did we? |
| 14:54.21 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:54.23 | hackeron | [TK]D-Fender: thats not incoming, I'm dialing a number from the phone |
| 14:54.24 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 14:54.38 | [TK]D-Fender | hackeron : you are dialing OUT on teliax <- |
| 14:54.57 | hackeron | yes |
| 14:55.09 | [TK]D-Fender | So where does that # lead to? |
| 14:55.25 | hackeron | [TK]D-Fender: analog phone in old office |
| 14:55.33 | [TK]D-Fender | On an analog line? |
| 14:55.44 | hackeron | [TK]D-Fender: the phone rings, someone picks it up, I cant hear them, they cant hear me |
| 14:55.55 | [TK]D-Fender | AH, sound issues |
| 14:56.09 | [TK]D-Fender | is your sever behind NAT? |
| 14:56.17 | hackeron | server isnt, phones are |
| 14:56.19 | santoshr | Whisk: http://pastebin.com/488696 |
| 14:56.54 | [TK]D-Fender | hackeron : Hmmmm, this is also IAX2 which is less susceptable to NAT problems.... |
| 14:57.14 | [TK]D-Fender | can you get audio to any number? |
| 14:57.19 | asteriskmonkey | i had a recent issue with iaxys like the one your talking about |
| 14:57.32 | Whisk | What extension are you logged on as? |
| 14:57.51 | asteriskmonkey | the problem was odd since its supposed to automatically traverse nat, resetting the cable/dsl router fixed the issue.. odd eh? |
| 14:58.10 | Whisk | i think you're looping it |
| 14:58.28 | [TK]D-Fender | hackeron : Tried from multiple phones on your side or did some quality testing with your current one to prove that audio is fine on it? |
| 14:58.32 | Whisk | Dial(Agent/12) calls the channel that agent 12 is logged onto |
| 14:58.33 | hackeron | [TK]D-Fender: so far, no -- I was able to before. Let me just revert my sip.conf file - I changed some stuff there :) |
| 14:58.49 | santoshr | i am loged in as agent 12 on extension 13 |
| 14:58.58 | Whisk | that needs to be a static extension, otherwise you'll just go round in circles |
| 14:59.02 | *** join/#asterisk sonic2wb (i=sonic2wb@user-11208d5.dsl.mindspring.com) |
| 14:59.16 | sonic2wb | Good Morning |
| 14:59.22 | santoshr | static extension |
| 14:59.26 | Whisk | you need an exten => 12,1,Dial(SIP/64) and to log on as 12 |
| 14:59.53 | Whisk | static as in not dialling an agent |
| 15:00.21 | santoshr | but wouldnt dial channel SIP. ? |
| 15:00.56 | santoshr | and call 64.. irrespective of whr 12 logged on as ? |
| 15:01.15 | Whisk | it would if you dial extension 12 |
| 15:01.39 | Whisk | but you're dialling 13 and the agent channel is forwarding it onto 12 (or wherever the agent is logged onto) |
| 15:02.11 | santoshr | but Whisk 12 is the agentID |
| 15:02.55 | hackeron | [TK]D-Fender: hmm, reverted config, no sound what so ever |
| 15:03.03 | Whisk | maybe 12 was a bad number for me to choose |
| 15:03.13 | [TK]D-Fender | hackeron : ugh. Does phone -> phone work? |
| 15:03.39 | Whisk | what i'm saying is that the agent must be logged on with an extension that points to a device, not the agent |
| 15:04.21 | Hmmhesays | ahh i love it when work screws me out of diner |
| 15:04.26 | Hmmhesays | *dinero even |
| 15:04.29 | hackeron | [TK]D-Fender: yes, phoning phone works |
| 15:04.50 | [TK]D-Fender | hackeron : Sounds like an ITSP problem then. |
| 15:05.38 | santoshr | no.. Whisk i dint get it..i am a little new to this.. can u please exemplify.. i would be grateul |
| 15:05.43 | santoshr | *grateful |
| 15:05.49 | hackeron | [TK]D-Fender: well, phoning phone's own extension rings the extension, but I cant hear myself or anything, let me get another phone, brb |
| 15:06.03 | Whisk | heh, sorry i'm not explaining this very well |
| 15:06.30 | hugo-v6 | hiho saftsack |
| 15:07.26 | Whisk | when you dial 13, it dials Agent/12, Asterisk then looks up where agent 12 is logged into, which in your case is extension 13@default. 13 then dials agent/12 which goes back to the beginning |
| 15:08.18 | hugo-v6 | q: i got bad echoes between * and sip-phones. i hear myself _really_ loud. is there a echocancelation which i could enable? |
| 15:08.47 | santoshr | ok.. but if I dial say (sip/64) then it would dial 64 irrespective of where 12 logged in |
| 15:09.21 | saftsack | hugo-v6, was it possible to apply the misdn patch to asterisk 1.2.1 too? |
| 15:10.28 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) |
| 15:10.28 | hackeron | [TK]D-Fender: no, I cant hear a thing if I phone an internal phone |
| 15:10.32 | hugo-v6 | saftsack: still not tried. but got no problems without it |
| 15:10.46 | hugo-v6 | at least here in the buero |
| 15:11.58 | saftsack | ok because i will patch it soon if this server is running here ;) |
| 15:13.32 | hugo-v6 | saftsack: well then tell me if the patch could be applied ;) thought u got the setup running? |
| 15:13.35 | Whisk | santoshr - look at http://pastebin.com/488723 - that's my config for this |
| 15:13.46 | *** join/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it) |
| 15:13.52 | saftsack | hugo-v6, ok i will |
| 15:13.55 | saftsack | is the patch big? |
| 15:14.12 | hugo-v6 | saftsack: nope. aprox 10-20 lines |
| 15:14.25 | agx | back... seems the problem is about the STUN server; anyone know a STUN server that will work 100% sure ? |
| 15:14.36 | [TK]D-Fender | hackeron : Guess you need to fix things on the inside then... |
| 15:14.59 | saftsack | ok so i can patch from hand if the patchfile isnt good for asterisk 1.2.1 |
| 15:15.10 | hackeron | [TK]D-Fender: hmm, wonder what was changed as I could talk on the phones before, hmmm |
| 15:15.12 | saftsack | or do you think a phone call to beronet would help? |
| 15:15.12 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:16.56 | santoshr | but Whisk would tht restrict agent 24 to log on only and only on device ed-1 ? |
| 15:17.03 | santoshr | i mena wouldnt |
| 15:17.32 | Whisk | no |
| 15:17.34 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 15:17.47 | [TK]D-Fender | hackeron : pastebin your sip.conf |
| 15:17.53 | hackeron | [TK]D-Fender: I've disabled both alsa and oss, could that have affected it? |
| 15:18.26 | [TK]D-Fender | hackeron : should have no impact |
| 15:18.32 | santoshr | yeah/.. sorry.. kinda got wht u were tyring to say.. |
| 15:18.39 | hugo-v6 | saftsack: concerning what? the patch? no dont think so. but if u want i can ask. i think ill call them tomorrow |
| 15:18.40 | santoshr | hold i will just be back.. trying wht u said |
| 15:18.55 | saftsack | ok that would be nice :) |
| 15:18.57 | saftsack | thanks |
| 15:19.24 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 15:19.54 | agx | Q: any public STUN server i can use that is currently working? |
| 15:19.54 | Ahrimanes | stun.foniristele.com should work |
| 15:19.54 | agx | Ahrimanes: thx |
| 15:20.31 | hackeron | [TK]D-Fender: yes, no effect :( |
| 15:20.55 | [TK]D-Fender | BBIAFM |
| 15:20.58 | hackeron | [TK]D-Fender: http://pastebin.com/488736 |
| 15:21.16 | hackeron | [TK]D-Fender: I trid to remove the nat=Yes also btw |
| 15:21.39 | RoyK | imho nat should be =yes |
| 15:21.54 | hackeron | RoyK: both in [general] and per extension? |
| 15:21.59 | RoyK | it only stops asterisk from caring about the IP in the SDP |
| 15:22.25 | RoyK | what's important, if not using a bunch of proxies etc, is not what's in the SDP, but what IP that is connecting |
| 15:22.55 | RoyK | using layer 3 addresses inside layer 7 packets is outright stupid |
| 15:23.17 | RoyK | IP addr inside SDP in ISP |
| 15:23.21 | RoyK | s/ISP/SIP/ |
| 15:23.31 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 15:23.54 | hugo-v6 | noone a hint or link to echocancelation with sip-phones? |
| 15:24.33 | RoyK | hugo-v6: most ATAs and sip phones have echo cancellation |
| 15:24.42 | Ahrimanes | echo usually only happens at pstn connections |
| 15:24.56 | RoyK | outgoing echo cancellation should be done in zaptel or hardware |
| 15:27.21 | hugo-v6 | royk: the phones should (at least with newfirmware) but there are still heavy echoes. i hear myself terribly loud. (the called party doesnt recognized it) |
| 15:27.42 | hugo-v6 | and only sometimes. not with every call i make |
| 15:27.46 | hackeron | [TK]D-Fender: any ideas? |
| 15:27.55 | santoshr | hey Whisk.. got things to work thanks alot. |
| 15:28.04 | santoshr | but hey i stil have a small issue.. |
| 15:28.04 | Whisk | heh nps :) |
| 15:28.06 | RoyK | hugo-v6: what clients? |
| 15:28.22 | hugo-v6 | RoyK: snom 190 phones |
| 15:28.28 | RoyK | ok |
| 15:28.30 | RoyK | no idea, then |
| 15:28.31 | RoyK | sorry |
| 15:28.35 | hackeron | [TK]D-Fender: PS, in case I havent mentioned it, the phones are behind a NAT in london, the server is in new york (not behind a NAT) |
| 15:28.44 | hugo-v6 | no problem. thank you anyway. |
| 15:28.57 | hugo-v6 | but echo cancellation from * side doesnt exist? |
| 15:29.30 | Ahrimanes | there's some options.. but hw is your best bet |
| 15:29.38 | [TK]D-Fender | hack, remove the externip from the phone config, set "canreinvite=no", set the "dtmfmode=rfc2833", and "nat=no" |
| 15:30.47 | hugo-v6 | hmm |
| 15:30.47 | hackeron | [TK]D-Fender: in [general] or per extension? |
| 15:30.50 | hugo-v6 | phone doesnt get any options concerning echo cancellation |
| 15:30.56 | santoshr | in ur case wht if u wanted 243 and 24 be the sames digits.. meaning i log in as 243 and if some one dials 243 it rings 243 |
| 15:31.06 | [TK]D-Fender | hackeron : and change the context in general to [incoming], and set the context in EACH phone to [myphones] |
| 15:31.19 | [TK]D-Fender | hackeron : the first batch was in the phone setups. |
| 15:31.30 | Cresl1n | hugo-v6: there isn't any echo cancellation done on IP to IP calls |
| 15:32.08 | Whisk | Well, you can't really do that - unless you have different contexts i spose |
| 15:32.43 | Whisk | What i've done is had the phone as e.g. 240 and the agent dial as 24 |
| 15:32.50 | Whisk | so agent logs in as exten 240 |
| 15:32.57 | *** join/#asterisk Modcuts (n=sam@82.133.98.155) |
| 15:33.01 | hugo-v6 | Cresl1n: this happens on ip-phones when calling into pstn. but on pstn-side i have echo-cancellation which works. beside the problem that i hear me on the sip-phone loud as hell sometimes |
| 15:33.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:33.23 | *** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no) |
| 15:33.32 | hackeron | [TK]D-Fender: ok, so like this? http://pastebin.com/488752 |
| 15:33.45 | saftsack | same prolbem here. on beginning calls i can hear myself too on the sip side |
| 15:33.51 | Cresl1n | hugo-v6: what version of * are you using? |
| 15:33.53 | santoshr | ok.. wht i wanted to do was remove the option of informing asterisk tht i am on this device. |
| 15:34.04 | saftsack | im telephoning with a budge tel 101 with alaw |
| 15:34.30 | *** join/#asterisk cnet2 (n=jjohn@200.122.157.91) |
| 15:34.59 | Modcuts | If you are trying to make a call with x-lite via asterisk and you get "All circits are currently busy, please try again later" at what point is the call failing? |
| 15:35.00 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 15:35.07 | Whisk | well, you can pass the extension to log in on as an option to agentcallbacklogin |
| 15:35.15 | *** join/#asterisk bkw__ (n=bkw_@adsl-69-148-35-86.dsl.tulsok.swbell.net) |
| 15:35.16 | hugo-v6 | Cresl1n: its 1.0.9 |
| 15:35.20 | santoshr | so the agentid becomes the extension |
| 15:35.35 | zoa | hey ho cresleke |
| 15:35.42 | santoshr | but hten there would be the same issue of dial tht i was facing.. correct |
| 15:35.57 | santoshr | it would go into a loop |
| 15:36.06 | Whisk | well, you'd still be logged in as e.g. 240, but you wouldn't ever know about it |
| 15:36.10 | cnet2 | hi, I have a digium card with 2Fxo and 2Fxs. I installed Asterisk@home, and configured it so that when i get a call from the pstn(fxo), it rings one of the extensions fxs. But for some reason there's a delay from the moment the call comes to the ringing of the extension of about 6 seconds, why can this be? |
| 15:36.46 | Whisk | but obviously you'd loose the ability to log in at a different location |
| 15:36.55 | santoshr | exactly.. |
| 15:37.14 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:37.26 | *** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net) |
| 15:37.50 | Whisk | let me show you a pastebin |
| 15:38.18 | santoshr | so tht would be soved if agentiD and the exten ur loggin into are the same.. but then how do u dial.. tht agent.. ? |
| 15:38.21 | santoshr | yeah ok |
| 15:38.28 | saftsack | build -> /usr/src/linux-2.6.8-24-obj/i386/default |
| 15:38.35 | saftsack | this folder doesnt exist :( |
| 15:38.36 | hackeron | [TK]D-Fender: that configuration in the pastebin won't work :( |
| 15:38.41 | saftsack | howto correct the link? |
| 15:39.07 | Cresl1n | hugo-v6: don't use 1.0.9 |
| 15:39.23 | Cresl1n | hugo-v6: try it with 1.2 (zaptel and asterisk) |
| 15:40.12 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 15:40.16 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:40.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:40.39 | Whisk | santoshr - look at http://pastebin.com/488763 |
| 15:40.43 | hugo-v6 | zaptel is no option here. because .de and ISDN :) i use mISDN. and there has nothing changed as far as i see, |
| 15:41.01 | [TK]D-Fender | hackeron : what are your 2 phones anda re they both local to the server? |
| 15:41.06 | Cresl1n | hugo-v6: Oh.... if you're using mISDN, I don't know |
| 15:41.13 | Cresl1n | hugo-v6: your mileage may vary |
| 15:41.14 | hackeron | [TK]D-Fender: no, they are in london, server is in new york |
| 15:41.21 | hackeron | [TK]D-Fender: phones are behind a nat, server isnt |
| 15:41.27 | [TK]D-Fender | are the phones behind nat where they are? |
| 15:41.34 | Cresl1n | hugo-v6: there have been a lot of changes in zaptel and chan_zap for echo issues since 1.0.9 |
| 15:41.38 | hackeron | [TK]D-Fender: yes, the phones are behind a nat |
| 15:42.13 | [TK]D-Fender | hackeron : ok, then add nat=yes on the phones, qualify=yes as well. |
| 15:42.25 | hugo-v6 | Cresl1n: i would love to hear something from the misdn ppl ;) |
| 15:42.40 | hackeron | [TK]D-Fender: and leave host=dynamic, right? |
| 15:42.45 | [TK]D-Fender | hackeron : yup |
| 15:42.51 | santoshr | Whisk: but then u have to have the same number of devices as the number of agents.. which would hugely remove scalability in my case |
| 15:42.56 | trym | notepad > asterisk |
| 15:43.09 | saftsack | where points the link in /lib/modules/KERNELVERSION/build ??? |
| 15:43.22 | hugo-v6 | santoshr: /usr/src/linux |
| 15:43.41 | hugo-v6 | s/linux/my-kernel-compile-dir/ |
| 15:43.41 | santoshr | hugo-v6: wht |
| 15:44.04 | hugo-v6 | santoshr: sorry, wrong nick |
| 15:44.09 | santoshr | ok |
| 15:44.22 | hugo-v6 | saftsack: s.o. |
| 15:44.32 | saftsack | ? |
| 15:44.36 | saftsack | ok |
| 15:44.51 | saftsack | thanks so its correct who i did |
| 15:44.52 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:44.56 | _Sam-- | how do i use the voicemail 'directory' command in extension.conf to have it play back the voicemailbox greetings of the voicemail users rather than have it ask for the first three letters of the last name |
| 15:45.01 | saftsack | but it doenst work :( |
| 15:45.21 | hugo-v6 | saftsack: this should normaly set from ur kernel-install |
| 15:45.27 | hugo-v6 | aehrm modules isntall |
| 15:45.45 | saftsack | yes i installed the kernel sources later |
| 15:45.50 | saftsack | (suse) *duck* |
| 15:45.58 | hugo-v6 | eyeyey |
| 15:46.02 | hackeron | [TK]D-Fender: no sound :'( |
| 15:46.05 | Modcuts | i just what to know what throws the "all circuits are currently busy" back when trying to ring a outside number? |
| 15:46.05 | hugo-v6 | use debian |
| 15:46.22 | saftsack | im using debian on all my systems but here is a buero what isnt mine |
| 15:46.35 | _Sam-- | Modcuts: probably your voip provider doesnt have enough capacity |
| 15:46.52 | _Sam-- | what you using to call the outside world? VOIP or PSTN? |
| 15:48.04 | saftsack | /bin/sh: scripts/basic/fixdep: No such file or directory |
| 15:48.04 | [TK]D-Fender | hackeron : repastebin |
| 15:48.31 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 15:48.31 | saftsack | ls /usr/src/linux/scripts/basic/ shows me fixdep.c |
| 15:48.36 | docelm0 | Hay [TK]D-Fender got a question for ya.. You got a sec? |
| 15:48.46 | hackeron | [TK]D-Fender: oh, I'm seeing notices like Jan 3 05:49:19 NOTICE[10028]: chan_sip.c:11328 sip_poke_noanswer: Peer '6272' is now UNREACHABLE! Last qualify: |
| 15:48.48 | santoshr | Whisk: dude any ideas ? |
| 15:49.05 | *** join/#asterisk __a (n=__a@85.105.12.111) |
| 15:49.10 | [TK]D-Fender | docelm0 : fire away |
| 15:49.20 | trym | not being able to reach something sucks |
| 15:49.21 | Modcuts | trying to use voiptalk |
| 15:49.27 | [TK]D-Fender | hackeron : not a good sign. Reboot the phone's while you're at it... |
| 15:49.29 | __a | guys, can asterisk register to multiple remote SIP proxies? |
| 15:49.56 | docelm0 | How many calls do you think asterisk could handle with a dual 3.2 and 2gb ram. Signaling only. No RTP transcoding.. Etc.. |
| 15:50.00 | __a | simultaneously i mean, like multiple register lines in sip.conf |
| 15:50.04 | Whisk | well, you wouldn't have to have the same number of devices, but you would have to have the same number of extensions, which would be a hardship |
| 15:50.27 | Whisk | because there's no reason you can't have an exten => 2X0,blah |
| 15:50.32 | *** part/#asterisk secure75 (n=mic@ppp-82-135-0-18.mnet-online.de) |
| 15:50.33 | Whisk | _2X0 even |
| 15:50.45 | Whisk | s/would/wouldn't |
| 15:50.59 | [TK]D-Fender | docelm0 : What interfaces? |
| 15:51.01 | hackeron | [TK]D-Fender: also seeing Jan 3 05:51:21 WARNING[10067]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'myphones' but I guess thats not too important |
| 15:51.28 | Modcuts | i have setup the trunk and the call routing i seems to work just gives me that, do you think it's the voip provider not asterisk? |
| 15:51.31 | [TK]D-Fender | hackeron : that shouldn't happen at ALL.... |
| 15:51.46 | [TK]D-Fender | hackeron : did you reboot the phones and watch them reregister? |
| 15:51.53 | hugo-v6 | saftsack: i dont know how u get such problems... nevver occured here. try a distribution (not a sandbox for kids) and vanilla kernel. u got lots of strange errors in the past. belive thats suse-specific |
| 15:51.53 | santoshr | no Whisk were are u going to with tht . |
| 15:52.17 | __a | [TK]D-Fender: do you know if it's possible to make asterisk to register to multiple proxies simultaneously? |
| 15:52.27 | saftsack | hugo-v6, yes i know that thats strange but i have to use suse |
| 15:52.33 | saftsack | but ill get a vanilla now |
| 15:52.33 | hackeron | [TK]D-Fender: I see Jan 3 05:52:56 NOTICE[10101]: chan_sip.c:11328 sip_poke_noanswer: Peer '7662' is now UNREACHABLE! Last qualify: 0 when I reboot the phones |
| 15:52.35 | docelm0 | SIP only |
| 15:52.39 | Whisk | not sure what you mean santoshr - perhaps you better explain exactly what you're trying to do |
| 15:52.56 | saftsack | http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.15.tar.bz2 |
| 15:53.01 | saftsack | :) |
| 15:53.09 | [TK]D-Fender | hackeron : Hmmm |
| 15:53.15 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 15:53.17 | [TK]D-Fender | __a : Sure |
| 15:53.20 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 15:53.43 | __a | [TK]D-Fender: any hints? |
| 15:53.51 | [TK]D-Fender | docelm0 : RTP pass-through or re-invite? |
| 15:54.17 | docelm0 | reinvite |
| 15:54.20 | hugo-v6 | saftsack: poor man. nobody should have to use suse |
| 15:54.30 | [TK]D-Fender | __a : Hints on what? Just do a "register => " for each service you want * to connect to and set up the incoming contexts. |
| 15:54.32 | saftsack | say this to my dad |
| 15:54.36 | santoshr | ok.. c in the current scenario thr are three parameters... agentid, pass, and the physical exten on which the user is loggin into.. wht i want is tht the agentid and the [hy exten be the same so tht it can be pased as a parameter to agentcallback .. but then how do i dial..? |
| 15:54.37 | docelm0 | Asterisk will only be doing the Signaling.. |
| 15:54.57 | hugo-v6 | saftsack: say to him. i wont do it with that crap of a distri |
| 15:55.00 | [TK]D-Fender | docelm0 : Dunno... thousands? Doesn't sound like much overhead at all. What service is * actually performing? |
| 15:55.04 | __a | [TK]D-Fender: you mean it's possible to have multiple register lines? |
| 15:55.07 | saftsack | :) |
| 15:55.14 | docelm0 | Load balancing |
| 15:55.29 | [TK]D-Fender | docelm0 : Keeping in mind * doesn't scale well in general. Sounds like you might want to use SER or something in front. |
| 15:55.33 | saftsack | i said him that we need a 4 port isdn card and not 4x 1port cards but he doesnt believe ... ^^ that will be fun |
| 15:55.35 | [TK]D-Fender | __a : Hell yeah. |
| 15:55.43 | hackeron | [TK]D-Fender: a google suggests grandstream phones have a problem with qualify=yes? |
| 15:55.46 | docelm0 | SER isnt doing the job. Its creating headaches for me cause when it gets a CANCEL it drops the cancel to the wrong machine not the machine that should get it. |
| 15:55.50 | docelm0 | I know asterisk wouldnt do this. |
| 15:55.56 | __a | [TK]D-Fender: COOL! |
| 15:56.08 | [TK]D-Fender | hackeron : if thats the case then you may be in trouble... not sure what to suggest.... |
| 15:56.12 | docelm0 | And I dont really need to scale that much.. Im hopeing for about 500 calls total (1000 channels) |
| 15:56.12 | zoa | grandstream phones have a problem with everything |
| 15:56.14 | docelm0 | tops |
| 15:56.25 | _Sam-- | i like my grandstream gxp200s |
| 15:56.28 | _Sam-- | 2000s |
| 15:56.35 | [TK]D-Fender | docelm0 : Could work I guess.... |
| 15:56.35 | _Sam-- | no problems with them on my stuff |
| 15:56.42 | docelm0 | _Sam--, I have 50 of them |
| 15:56.51 | _Sam-- | i have only 15 |
| 15:56.58 | _Sam-- | ocassionally they will lock up |
| 15:56.59 | hackeron | _Sam--: thats the ones we're using. Any chance I can have a look at your configuration? |
| 15:56.59 | [TK]D-Fender | docelm0 : Scripts your configs for anti-carpal-tunnel ! |
| 15:57.04 | _Sam-- | but i cant complain |
| 15:57.10 | _Sam-- | what is your problem with them? |
| 15:58.01 | docelm0 | anti what? huh? |
| 15:58.04 | hackeron | _Sam--: I can phone and accept calls but there's no sound coming in or out |
| 15:58.24 | zoa | nat problem, nat problem, nat problem |
| 15:58.25 | docelm0 | hackeron, check NAT/Stund |
| 15:58.30 | docelm0 | err Stun |
| 15:58.52 | _Sam-- | zoa: how do i make the voicemail 'directory' command play back user greetings instead of asking for first 3 digits of last name |
| 15:59.01 | hackeron | docelm0: they were working before without stund - not sure what I changed to stop them working :( |
| 15:59.04 | [TK]D-Fender | docelm0 : "carpal-tunnel-syndrome" Also known as RSI (repetative stress injury). sounds like you're setup is large enough to cause cramps :) |
| 15:59.19 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
| 15:59.23 | *** join/#asterisk alvariux (n=unky@dsl-201-129-81-130.prod-infinitum.com.mx) |
| 15:59.27 | alvariux | hello |
| 15:59.28 | santoshr | Whisk: dude...u around |
| 15:59.39 | docelm0 | Well just until the customer base is more stable then we will be moving to a more commercial application asterisk will be a stepping stone for right now |
| 15:59.55 | alvariux | anybody can help with my voicemail |
| 16:00.01 | hugo-v6 | how can i increase asterisk logging? |
| 16:00.04 | Whisk | yer |
| 16:00.09 | ast_freak | alvariux, what's wrong with it? |
| 16:00.12 | docelm0 | hugo-v6, logger.conf |
| 16:00.23 | hugo-v6 | thank you docelm0 |
| 16:00.33 | santoshr | Whisk: >>> ok.. c in the current scenario thr are three parameters... agentid, pass, and the physical exten on which the user is loggin into.. wht i want is tht the agentid and the [hy exten be the same so tht it can be pased as a parameter to agentcallback .. but then how do i dial..? <<<< |
| 16:00.45 | alvariux | i can record messages but when i try to check it i havin a segmentation fault /usr/sbin/safe_asterisk: line 42: 2296 Segmentation fault ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} |
| 16:01.19 | ast_freak | alvariux, what version are you using? |
| 16:01.31 | alvariux | 1.2.1 |
| 16:01.57 | zoa | sam, i dont know if thats possible |
| 16:02.15 | _Sam-- | so if the caller doesnt know the person's last name, they cant find out their voicemailbox? |
| 16:02.21 | ast_freak | alvariux, how about some verbose info from the console while this is happening? |
| 16:02.32 | Whisk | you dont santoshr |
| 16:02.43 | zoa | want us to have a look at it tomorrow ? |
| 16:02.45 | santoshr | meaning.. |
| 16:02.46 | Whisk | have the physical extension as agentid + something |
| 16:02.46 | _Sam-- | i could just make a "background" to play it |
| 16:02.54 | alvariux | Executing VoiceMailMain("SIP/102-a522", "") in new stack |
| 16:02.54 | alvariux | <PROTECTED> |
| 16:02.54 | alvariux | <PROTECTED> |
| 16:02.54 | _Sam-- | and have it do the right things via menuing |
| 16:02.56 | Whisk | and strip the something off to pass to the agentcallbacklogin |
| 16:03.16 | santoshr | hmm |
| 16:03.19 | alvariux | ast_freak when i type the mailbox happens that |
| 16:03.30 | zoa | sam, true |
| 16:03.47 | ast_freak | alvariux, when you type the first digit, or a couple? |
| 16:04.04 | alvariux | i think the whole mailbox |
| 16:04.11 | alvariux | that would be 102 |
| 16:04.21 | saftsack | the vanilla kernel. some piece linux in my suse system *gg* |
| 16:04.29 | ast_freak | alvariux, Change the language back to en and see what happens. |
| 16:04.49 | *** join/#asterisk manolo (n=manolo@200.124.172.72) |
| 16:05.19 | hugo-v6 | saftsack: if ou use patches like misdn neverever try to use an suse-kernel |
| 16:05.41 | saftsack | ok :) |
| 16:05.48 | manolo | Hi pals, got new trouble today: |
| 16:05.59 | hugo-v6 | i never use distro kernels. i dont use debian-kernels either |
| 16:06.14 | saftsack | you dont have to say that to me ;) |
| 16:06.18 | saftsack | i do that too, at home |
| 16:06.24 | fugitivo | hugo-v6: i don't use distros |
| 16:06.39 | saftsack | lfs? |
| 16:06.43 | fugitivo | yes |
| 16:06.46 | saftsack | :) |
| 16:06.55 | saftsack | alfs or normal lfs? |
| 16:07.09 | fugitivo | normal lfs |
| 16:07.14 | alvariux | ast_freak is the same -- SIP Seeding peer from astdb: '102' at 102@201.129.81.130:9638 for 3600 |
| 16:07.14 | alvariux | <PROTECTED> |
| 16:07.14 | alvariux | <PROTECTED> |
| 16:07.14 | alvariux | <PROTECTED> |
| 16:07.14 | alvariux | bisteck*CLI> /usr/sbin/safe_asterisk: line 42: 2332 Segmentation fault ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} |
| 16:07.16 | alvariux | Asterisk ended with exit status 139 |
| 16:07.18 | saftsack | fugitivo, do you have automated update scripts? |
| 16:07.18 | alvariux | Asterisk exited on signal 11. |
| 16:07.20 | fugitivo | with some modifications |
| 16:07.20 | [TK]D-Fender | What's "alfs"? |
| 16:07.28 | saftsack | automated lfs |
| 16:07.35 | hugo-v6 | fugitivo: too much work ;) debian works fine for me. but the kernels not ;) |
| 16:07.39 | saftsack | it is a script set for installing lfs without a hand on the tastature |
| 16:07.44 | *** join/#asterisk seele_ (n=seele@200.124.172.72) |
| 16:07.50 | fugitivo | saftsack: what's the fun of that? :) |
| 16:07.57 | [TK]D-Fender | saftsack : a quick build script? I've been thinking about LFS for a while coming from Slackware.... |
| 16:07.59 | manolo | I have two TDM4000 with 3 FXO and 5 FXS, in total, I got my outcoming calls getting out of the three FXO all right, but, my incoming calls are only getting by one FXO.. so it gets busy tone |
| 16:08.00 | fugitivo | i cut it to fit a 128mb compact flash |
| 16:08.02 | ast_freak | alvariux, sorry, this is beyond me. :^( |
| 16:08.04 | saftsack | fugitivo, do you mean alfs or update script? |
| 16:08.11 | fugitivo | saftsack: alfs |
| 16:08.17 | saftsack | [TK]D-Fender, i had lfs and it was the fastest system ever :) |
| 16:08.22 | saftsack | ill give it a try too :) |
| 16:08.33 | alvariux | ast_freak thanks |
| 16:08.34 | saftsack | fugitivo, yes thats true but sometimes for lazy people :) |
| 16:08.58 | [TK]D-Fender | I don't care about "fastest". I care about stable, reliable, SANE, and hopefully "convenient". A tough request I know. |
| 16:09.16 | [TK]D-Fender | I may be better off with a Debian varient.... |
| 16:09.20 | fugitivo | [TK]D-Fender: it's fast and stable, no crap inside |
| 16:09.51 | hugo-v6 | d-fender: knoppix *duck&run* |
| 16:10.04 | ErMeS|Work | DEBUG[1479] chan_sip.c: That's odd... Got a response on a call we dont know about. |
| 16:10.20 | [TK]D-Fender | hugo-v6 : Ummm, I found it reather ugly. ANd I'd rather use a distro with all the devel stuff included. |
| 16:10.31 | fugitivo | saftsack: for updates, i have a host system, compile everything there, then a script will copy the files to the servers |
| 16:10.32 | ErMeS|Work | NOTICE[1479] pbx.c: Cannot find extension context 'from-internal' |
| 16:10.41 | iCEBrkr | That's simple |
| 16:10.49 | saftsack | fugitivo, do you have a serverpark? :) |
| 16:11.05 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
| 16:11.09 | fugitivo | saftsack: what do you mean with serverpack? |
| 16:11.18 | fugitivo | park |
| 16:11.21 | saftsack | are you an admin in company? |
| 16:11.44 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
| 16:11.47 | hugo-v6 | d-fender: was a joke ;) i would never recommend that unless u need a linux on a hw u dont know (ie gf new pc) |
| 16:11.56 | fugitivo | yes, and i have several servers in other companies |
| 16:12.21 | [TK]D-Fender | hugo-v6 : I'd rather use Mepis :) |
| 16:12.36 | hugo-v6 | swm: damn.. would love to smoke one too now |
| 16:12.44 | saftsack | fugitivo, cool :) |
| 16:12.45 | iCEBrkr | swm_: Obviously you got money to throw away. |
| 16:12.48 | hugo-v6 | mepis? |
| 16:12.52 | saftsack | that would be a good jo |
| 16:12.52 | saftsack | b |
| 16:12.57 | manolo | Also, when i dial my own number. from inside extensions.. the main incoming route gets blocked, like in some kind of "loop" |
| 16:12.59 | hugo-v6 | thx swm :) |
| 16:13.16 | alvariux | anybody can help me with my voicemail |
| 16:13.41 | manolo | So in order to override that i have to unplug the line directly from the cards. |
| 16:14.31 | *** part/#asterisk __a (n=__a@85.105.12.111) |
| 16:14.32 | hugo-v6 | swm: when they are ready leave me a msg :p |
| 16:14.37 | manolo | Hello any help with that, please anyone readn my previous posts |
| 16:14.40 | [TK]D-Fender | hugo-v6 : Never heard of? Its a very popular Debian derivative that runs the std repos, etc and installs from a live CD at the same time. Its very highly rated on Distrowatch |
| 16:15.05 | hugo-v6 | d-fender never heared of. but ill have a look atm |
| 16:15.36 | swm_ | I think digium needs to create a feature request board and put requested features on it so people can say I want asterisk to do this and that and everyone vote so we can focus the development of asterisk :) |
| 16:16.04 | [TK]D-Fender | hugo-v6 : its #5 on Distrowatch - 5 MEPIS 939 |
| 16:16.16 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 16:16.22 | [TK]D-Fender | http://distrowatch.com/table.php?distribution=mepis |
| 16:16.29 | manolo | HELLO!!! |
| 16:16.33 | swm_ | So the more people that vote for certian features will increase the value of the feature thus digium could create some kind of credit program for whoever developes the feature and in return give the developer credits :) |
| 16:17.26 | [TK]D-Fender | swm_ : A horse built by commitee......*shudder*. Not sure it'd work with an animal like *... |
| 16:18.12 | swm_ | never know till you try! |
| 16:18.43 | hugo-v6 | ubuntu on 1st place suse on 3rd... what a world |
| 16:18.56 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc11a.dialup.mindspring.com) |
| 16:19.22 | Cresl1n | swm_: that's kind of what the bounty board is for |
| 16:20.10 | swm_ | yeah but ... Digium could give people credits instead and they could get free t-shirts tdm cards if they work for 10 years making code heh and stuff lol |
| 16:20.28 | eKo1 | hah |
| 16:20.29 | [TK]D-Fender | hugo-v6 : SUSE lost their following when Novell took over... and Ubuntu is good for newbs which have us outnumbered :) |
| 16:20.33 | eKo1 | that'll happen |
| 16:20.45 | swm_ | ideas |
| 16:20.46 | swm_ | heh |
| 16:20.48 | [TK]D-Fender | I like certain parts of Ubuntu, but without a normal root account I feel like an idiot! |
| 16:21.34 | [TK]D-Fender | So for now I'm stuck with Slack.... |
| 16:21.51 | fugitivo | ubuntu is crap |
| 16:22.09 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:22.36 | fugitivo | [TK]D-Fender: why you want to move from slack? |
| 16:22.39 | hugo-v6 | d-fender: well im searching for a lightweight distro for my gf's old box... maybe ill give ubuntu a try |
| 16:22.51 | saftsack | take lfs :) |
| 16:23.01 | ErMeS|Work | iCEBrkr |
| 16:23.06 | hugo-v6 | no way. she have to use it ;) |
| 16:23.15 | saftsack | ok ... ^ |
| 16:23.18 | ErMeS|Work | NOTICE[1479] pbx.c: Cannot find extension context .... |
| 16:23.27 | saftsack | the waf is a important thing |
| 16:23.36 | fugitivo | hugo-v6: knoppix |
| 16:23.44 | zoa | http://www.asteriskguru.com/tutorials/cannot_find_extension_context.html |
| 16:24.08 | hugo-v6 | fugitivo: nope. ill think shell get damn small linux |
| 16:24.41 | [TK]D-Fender | fugitivo : I need to learn SysV init's, I'd like to be able to use more standard packaging for the "boring" stuff (like GUI & related tools). |
| 16:24.47 | fugitivo | hugo-v6: whhy not? |
| 16:24.55 | hugo-v6 | but i have to find a webcam solution for her, so we can do a lil bit of cam-chat :p |
| 16:25.04 | [TK]D-Fender | fugitivo : which is why I think I may be best served by a Debian derivative. |
| 16:25.10 | hugo-v6 | fugitivo: to much hdd-space wasted. |
| 16:25.11 | fugitivo | [TK]D-Fender: "packaging" is not standard |
| 16:25.24 | [TK]D-Fender | hugo-v6 : Ubuntu is NOT light :) |
| 16:25.26 | fugitivo | hugo-v6: it's only a cd |
| 16:25.45 | [TK]D-Fender | SLAX is light :) |
| 16:26.00 | hugo-v6 | fugitivo: i knw. i have it here. but dsl will use only a 13 or less of knoppix. |
| 16:26.23 | fugitivo | hugo-v6: then install lfs with the packages you need :) |
| 16:26.42 | hugo-v6 | d-fender: saw that and have a look on dsl now ;) |
| 16:27.09 | hugo-v6 | fugitivo: too much work. but maybe this alfs will do it. |
| 16:27.21 | *** join/#asterisk freestyle_networ (n=chatzill@68.148.192.184) |
| 16:27.53 | fugitivo | hugo-v6: or go with gentoo |
| 16:28.03 | fugitivo | hugo-v6: just emerge what you need |
| 16:28.06 | [TK]D-Fender | DSL? As in Damn Small Linux? Cute but... ICK. WOuld never use it installed on a desktop, only as a portable OS. |
| 16:28.09 | [TK]D-Fender | y0 |
| 16:28.35 | ManxPower | I don't suppose anyone has Polycom SIP 1.6.3? there's a feature in there I want to use. |
| 16:28.35 | hugo-v6 | well... now im going to get cigarettes and when im back a drink |
| 16:28.36 | tzafrir_laptop | What's that "old" box? |
| 16:28.37 | freestyle_networ | anyone here knowledgeable on ztdummy? |
| 16:28.39 | fugitivo | hugo-v6: if it's for your gf, install something like knoppix, not dsl |
| 16:28.49 | [TK]D-Fender | ManxPower : I do. |
| 16:28.54 | tzafrir_laptop | Ubntu means gnome, with very little light-wieght alternatives |
| 16:29.04 | ManxPower | [TK]D-Fender, can you send? If so, what's the best way? |
| 16:29.06 | hugo-v6 | fugitivo: she got an old desktop of mine. that thing would compile on gentoo over a month i guess |
| 16:29.11 | [TK]D-Fender | tzafrir_laptop : XFCE :) |
| 16:29.22 | [TK]D-Fender | ManxPower : DCC or FTP. |
| 16:29.32 | tzafrir_laptop | Knoppix (or any live CD) means wasting precious memory on a ramdisk. |
| 16:29.42 | hugo-v6 | fugitivo: well ill give a few a try and she should choose the one she likes |
| 16:29.45 | ManxPower | [TK]D-Fender, can you put it somewhere I can grab it from? |
| 16:30.06 | seele_ | [TK]D-Fender, to get a normal root account in ubuntu try this: sudo passwd root |
| 16:30.10 | ErMeS|Work | zoa |
| 16:30.16 | ManxPower | [TK]D-Fender, have you used this feature " 12761: Added support for setting flash parameters from configuration file" |
| 16:30.19 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
| 16:30.55 | zoa | yes ErMeS|Work ? |
| 16:31.00 | hackeron | I dont get it, I have nat=yes and externip=theip for extensions, but I still have no audio incoming or outgoing -- If I connect same phone to a voip provider, it works just fine. Any idea what could be wrong? (the phones behind a NAT, the asterisk server isnt) |
| 16:31.05 | [TK]D-Fender | ManxPower : Never heard of. Whats it for? |
| 16:31.18 | ErMeS|Work | context "must" be in extensions.con or it can be also in another file and in extensions.conf : #include another_file ? |
| 16:31.30 | ManxPower | [TK]D-Fender, Sounds like I can enable CDP via a config file on the server, which would ROCK |
| 16:31.36 | [TK]D-Fender | CDP? |
| 16:31.39 | ManxPower | hackeron, and you have localnet= too. |
| 16:31.44 | ManxPower | [TK]D-Fender, VLAN auto discovery |
| 16:31.52 | Seldon1975 | has anyone here actually observed the Comedian mail terminating voicemail recording after {maxmessage} |
| 16:32.01 | Seldon1975 | it doesnt seem to be working at my site |
| 16:32.04 | zoa | ErMeS|Work: you can also include it, yes |
| 16:32.05 | [TK]D-Fender | ManxPower : Hosting it for you now. It'll be mounted in a few... |
| 16:32.13 | hackeron | ManxPower: how's that going to help? the phones are not on the local network |
| 16:32.13 | ManxPower | [TK]D-Fender, Thanks! |
| 16:32.37 | ManxPower | hackeron, Huh? localnet= tells asterisk what the local network is, anything not local it will do special NAT for. |
| 16:32.58 | ErMeS|Work | i included but it doesn t work |
| 16:33.06 | ManxPower | works with externip= (which should be the public IP of the nat router asterisk is connected to) |
| 16:33.58 | ManxPower | EriSan, #include merges the files BEFORE extensions.conf is processed. |
| 16:34.07 | ManxPower | JUST like a compiler does. |
| 16:34.38 | _Sam-- | i dont use it...but is AMP fully featured enough to configure a relatively complex pbx? |
| 16:34.54 | hackeron | ManxPower: hmm, that worked! -- its very, very crackely, but I can hear stuff, thanks! |
| 16:34.57 | _Sam-- | or it is still easier to do some things by hand? |
| 16:35.10 | EriSan | ManxPower, guess you meant ErMeS|Work |
| 16:35.34 | ManxPower | EriSan, Ya'll are just a sea of newbies to me. |
| 16:35.47 | EriSan | ;) |
| 16:36.20 | manolo | hey, whats the key for taking an outside call thats ringing in another extension?? |
| 16:36.30 | docelm0 | Who you callin a newb |
| 16:36.31 | wasim | pickup groups |
| 16:36.34 | docelm0 | :) |
| 16:36.41 | manolo | Phone 1 is ringing and i want to take that call from Phone 2? |
| 16:36.56 | santoshr | Whisk: we did something like this.. problem solved.. AgentCallbackLogin(${test},,${CALLERIDNUM}@testing) |
| 16:36.58 | zoa | _Sam--, amp can do most of it |
| 16:37.08 | zoa | but i dont like it |
| 16:37.15 | santoshr | so the agent can login from any device. |
| 16:37.17 | ManxPower | manolo, see "pickup" in the Wiki |
| 16:37.21 | _Sam-- | me either, its not intuitive at all |
| 16:37.46 | g__ | Question: was echo cancellation improved between 1.0.9 and 1.2.x? |
| 16:37.48 | _Sam-- | i think there should be an initial "wizard" type process to setup the first time |
| 16:38.20 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 16:38.21 | g__ | Followup question: does a hardware echo canceller work better than the software one? |
| 16:38.44 | wasim | g__: they are all software! |
| 16:39.07 | g__ | microcode on a dsp? |
| 16:39.32 | fugitivo | wasim: that's not true |
| 16:39.32 | *** join/#asterisk Strom_C (n=strom@198.172.114.2) |
| 16:39.38 | Strom_C | yo |
| 16:39.47 | fugitivo | good hardware has echocan :) |
| 16:40.11 | Strom_C | is there a zaptel command-line utility that will tell me the alarm status on the four T1 spans? |
| 16:40.14 | benjk | wasim: if you drill the two wires together an create a shortcut, that'll be a haardware echo canceller |
| 16:40.21 | g__ | fugitivo: but is hardware echocan better than software echocan, or is just easier on your cpu cycles? |
| 16:40.24 | saftsack | make oldconfig on a suse system is a horror |
| 16:40.39 | wasim | benjk: i was just thinking on how it would work on a tincan with string protocol :) |
| 16:40.47 | fugitivo | g__: hardware is always better than software |
| 16:40.54 | saftsack | compiling started 30minutes ago |
| 16:40.57 | saftsack | on a fast system |
| 16:41.00 | benjk | it will cancel the signal too, but anyway ... |
| 16:41.30 | fugitivo | Strom_C: zap show status from the cli |
| 16:41.49 | g__ | fugitivo: is there a good reason for that? I'm all in favour of dedicated and specialized hardware, though.. |
| 16:42.10 | [TK]D-Fender | g__ : What kind of interface are you looking to use? |
| 16:43.35 | Strom_C | should the zaptel card automatically respond to a request for a loopback? SBC is trying to run a test on my circuit and they're claiming my CSU isn't responding to a loopback test... |
| 16:44.21 | g__ | We're using a Wildcard TE110P and we're having occational echo problems.. trying to decide if buying a TE411P would solve the problem.. |
| 16:44.28 | *** join/#asterisk Defraz (n=t0tal@103-16.69-92-cpe.cableone.net) |
| 16:44.54 | Seldon1975 | has anyone here actually observed the Comedian mail terminating voicemail recording after {maxmessage}; it seems not to be working at my site; maxmessage=600 but someone left a 28 minute message |
| 16:44.58 | [TK]D-Fender | g__ : how many ports do you need? |
| 16:45.06 | iDunno | 63.5 million. |
| 16:45.07 | g__ | Just 1, right now. |
| 16:45.54 | [TK]D-Fender | g__ : If it had been more I'd say jump right over to the Sangoma A104d. I haven't heard echo but ONCE since its install. (which I'm sure was HIS fault :)) |
| 16:46.03 | Druken | Seldon1975: how in hell does someone leave a 28 min message? |
| 16:46.19 | [TK]D-Fender | g__ : They will be coming out with lower density versions soon though. |
| 16:46.30 | Strom_C | I've had no echo problems with my TE406P |
| 16:46.44 | Seldon1975 | Druken: well I assume that the line just stayed up |
| 16:46.54 | Seldon1975 | Druken: because * didnt hang up the line |
| 16:47.10 | Druken | Seldon1975: well, it should term the recording after 3-5 seconds of silence |
| 16:47.23 | Seldon1975 | yes, it should |
| 16:47.30 | Seldon1975 | it seems not to do so |
| 16:47.36 | g__ | [TK]D-Fender: did you notice a difference switching from software-only echocan to hardware? I mean, we noticed a huge difference going from analogue to digital.. but there's still that residual problem. |
| 16:47.38 | Seldon1975 | has anyone else observed such an issue? |
| 16:47.44 | Druken | perhaps you have noisy lines? |
| 16:47.53 | file | okay |
| 16:48.04 | file | no matter how many times I say "Colp - P as in Peter" nobody can get my last name right over the phone |
| 16:48.18 | Seldon1975 | Druken: do you mean so that the silence period wasnt detected? It should still have terminated after 10 minutes, no? |
| 16:48.55 | [TK]D-Fender | g__ : the S/W one royall sucked ass for me and running it in hardware on my A104d there isn't *ANY* |
| 16:48.56 | Druken | Seldon1975: probably.. did you RESTART after you changed the config? |
| 16:49.02 | _Sam-- | manx did you get the dtivo on ebay? |
| 16:49.19 | Seldon1975 | Druken: yes, the box itself has even been restarted several times |
| 16:50.00 | Seldon1975 | Druken: I think there's a (possibly hardware) problem which prevents the card from hanging up the line |
| 16:50.12 | _Sam-- | you have to replace the hard drive in your new tivo with a hacked one...(or hack the one in there, but thats riskier)....then all your tv recordings are unencrypted mpg, and you have TCP/IP and can transfer them / play them from the tivo to other devices |
| 16:50.13 | *** join/#asterisk anonymouz666 (n=anonymou@gw.ideavalley.com.br) |
| 16:50.31 | Druken | Seldon1975: what kinda hardware? |
| 16:50.56 | Seldon1975 | Druken: its one of the new TDM2400E cards with 2 quad FXO modules and two Quad FXS |
| 16:51.23 | Druken | oh well shit.... i'm glad SOMEONE has money.... |
| 16:51.55 | Seldon1975 | Druken: lol |
| 16:52.22 | Seldon1975 | Druken: I was really just wondering if others had seen the voicemail program successfully hang up so I can dismiss that software as a possible culprit |
| 16:52.38 | Seldon1975 | i mean hang up after the timeout |
| 16:52.42 | Seldon1975 | it's not a known issue? |
| 16:52.44 | Druken | well, i personally have never played with it... |
| 16:53.01 | *** join/#asterisk kiwnix (n=egarcia@201.red-82-158-154.user.auna.net) |
| 16:53.08 | Druken | i hate timed messages... course, i hate 20 - 40 second timed messages... |
| 16:53.31 | *** join/#asterisk iPBX (n=owned@68-169-204-147.agstme.adelphia.net) |
| 16:53.53 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 16:54.00 | freestyle_networ | anyone work with timming devices for conference calling? |
| 16:54.47 | ErMeS|Work | whichever context i write in sip.conf ... it goes on to tell... cannot fin context .... |
| 16:54.56 | cnet2 | when i make a call to my asterisk box, through the zaptel fxo.. if i get no answer and hangup, the zaptel idles.. what can this be? |
| 16:55.17 | zoa | ErMeS|Work: then there is some typo in the config file |
| 16:55.22 | cnet2 | i'm using asterisk@home |
| 16:55.27 | zoa | read the thing i sent you again |
| 16:55.43 | *** join/#asterisk justinu (n=justinu@207.181.0.86) |
| 16:56.23 | cnet2 | me ? |
| 16:56.33 | alvariux | ast_freak have you used voicemail with realtime asterisk? |
| 16:56.53 | ErMeS|Work | i read it a lot of times |
| 16:57.09 | ErMeS|Work | this time i spent it to find out somethiung wrong |
| 16:57.27 | zoa | start with the default config |
| 16:57.51 | cnet2 | zoa: talk'n to me ? jeje :S |
| 16:58.03 | zoa | hmm no |
| 16:58.11 | zoa | it was to ErMeS|Work |
| 16:59.36 | alvariux | anybody have you used voicemail with realtime asterisk? |
| 17:04.06 | *** join/#asterisk Firebird_ (n=xxx@130.40.39-62.rev.gaoland.net) |
| 17:04.20 | *** join/#asterisk razu_ (n=razu@ip59.cab62.mus.starman.ee) |
| 17:08.16 | g__ | Do you know what Sangoma's more expensive than Digium? |
| 17:08.30 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 17:08.34 | PakiPenguin | evening |
| 17:08.34 | g__ | (That was for [TK]D-Fender) |
| 17:08.38 | PakiPenguin | is the svn repos. down? |
| 17:08.42 | g__ | morning! |
| 17:08.48 | LoRez | g__: they have DSPs on board |
| 17:08.54 | PakiPenguin | i am getting this svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 17:08.54 | PakiPenguin | svn: Caught signal |
| 17:08.54 | LoRez | (probably) |
| 17:09.05 | ast_freak | alvariux, no sir. |
| 17:09.22 | PakiPenguin | can i download the code from somewhere else? |
| 17:09.30 | g__ | LoRez: why? (Or why doesn't Digium's cards have them?) |
| 17:10.04 | LoRez | g__: digium's cards don't have them because it handles all that in software. makes the cards cheaper. |
| 17:10.16 | alvariux | ast_freak i found the problem, is not getting the mailbox |
| 17:10.29 | alvariux | if i have a static conf in voicemail.conf it works |
| 17:10.38 | alvariux | but im using realtime asterisk |
| 17:11.06 | [TK]D-Fender | g__ : Thats because Sangoma's use a real DSP and its specs DESTROY Digium's (look at the taps/channels rate). Also Sangoma cards work on either PCI voltage, and play nicely with interrupts and do a lot more in hardware. |
| 17:11.27 | justinu | sangoma cards aren't much more |
| 17:11.30 | justinu | i think mine was 25 bucksm ore |
| 17:11.30 | [TK]D-Fender | g__ : And run on Windows as well. |
| 17:11.42 | g__ | That's a feature? |
| 17:12.18 | [TK]D-Fender | g__ : YES. As much as I love Linux and *, I don't want my hardware to "own" me. EVER. |
| 17:12.26 | [TK]D-Fender | Freedom!!! |
| 17:13.07 | g__ | Oh, I'm all for that.. |
| 17:13.10 | PakiPenguin | justinu, where did you get your sangoma card from? |
| 17:13.46 | g__ | Perhaps you ment "And it runs on FreeBSD as well". |
| 17:13.54 | justinu | i bought mine from atacomm, iirc |
| 17:14.05 | [TK]D-Fender | g__ : that too, but it has native Windows drivers as well. |
| 17:15.31 | [TK]D-Fender | (not that I've had any reason to use it) |
| 17:15.44 | freestyle_networ | We had good luck with Sangoma before moving to DS's ..we found that they start to loose timing over 3 PRI's with a dual xeon |
| 17:15.56 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 17:15.58 | justinu | i just got an email from pulver.com |
| 17:16.09 | justinu | apparently the new FWD client will support SIP and IAX |
| 17:16.24 | g__ | What do you mean by "loose timing"? |
| 17:16.37 | asterisk99 | anyone here using the Digium IAXy phone adapter? I am having a wierd sound problem with mine |
| 17:16.40 | Strom_C | i think he means "lose timing" |
| 17:16.47 | freestyle_networ | start to get call artifacts,an IRQ issue |
| 17:16.54 | Strom_C | remember kids, loose and lose are TWO DIFFERENT WORDS |
| 17:17.03 | *** join/#asterisk dcoulson (n=dcoulson@wilbur.geekcolony.net) |
| 17:17.13 | freestyle_networ | but yeah, Sangoma hands down if your doing PRI's ... i wouldnt recommend their new DS3 card, i would never be that brave |
| 17:17.55 | justinu | that doesn't sound right |
| 17:17.55 | [TK]D-Fender | freestyle_networ : that and its not cahnnelized for voice IIRC |
| 17:17.57 | g__ | I suppose it depends what kind of Xeons.. |
| 17:18.14 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 17:18.20 | freestyle_networ | we had echo cancellation turned on, which beats up the server good |
| 17:18.39 | freestyle_networ | they were dual xeons, its really a function of interupts not CPU bandwidth |
| 17:18.45 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
| 17:18.51 | Strom_C | do the sangomas do hardware echo cancellation? |
| 17:18.58 | [hC] | morning! |
| 17:19.10 | [hC] | Strom_C: yup, 128ms |
| 17:19.12 | *** join/#asterisk shodan (n=shodan@ip010.99-113-216.pppoe4.joliette.intermonde.net) |
| 17:19.12 | freestyle_networ | g__ yeah it hangled it, anything past 3 PRI's was uncharted territory for call quality |
| 17:19.44 | [hC] | Anyone using Linksys SPA-941's in here? |
| 17:20.03 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 17:20.04 | g__ | Ok.. thanks for the feedback. Sounds like it's worth the extra $$ |
| 17:20.05 | freestyle_networ | yes they do, we were involved in testing their new drivers that moved echo cancellation from zaptel to their WANPipe drivers, i would recmmonend them over Digium crap anyday, they actually give you support techs that give a crap |
| 17:20.17 | shodan | 25$ per fxs, is it there yet ? |
| 17:20.18 | g__ | I'll see what I can swing. |
| 17:20.36 | rajiv|work | anyone here using T1 and/or TE110P with * ? |
| 17:20.41 | freestyle_networ | Yeah Digium is happy too help and get user feedback |
| 17:20.43 | [hC] | I quite like my sangoma a102u |
| 17:20.46 | justinu | digium doesn't give a crap? say it isn't so |
| 17:20.47 | freestyle_networ | oops, Sangoma i meant! |
| 17:21.09 | freestyle_networ | yeah dont get me started on Digium ..we only use * now for Voicemail and conference calling |
| 17:21.28 | rajiv|work | i am looking to setup a new site with asterisk. they have 4 analog lines and dsl service now. would it be easier to move them to T1 service and do everything through a linux box ? |
| 17:21.36 | [hC] | what do you use for call routing and sip registration? |
| 17:21.53 | justinu | just going to astricon and trying to talk to them, i could tell they didn't give a crap |
| 17:22.07 | justinu | at least I learned quickly who NOT to call for help :) |
| 17:22.29 | freestyle_networ | we were using SER ..but have recerntly moved to a 7th generation carrier grade switch ...Metaswitch, 12 DS3 capability |
| 17:22.35 | shodan | what's the cheapest per fxs in the 5-10-20 port range ? |
| 17:22.56 | justinu | freestyle_networ: sounds expensive |
| 17:23.17 | PakiPenguin | freestyle_networ, which vendor? |
| 17:23.17 | [hC] | freestyle_networ: ah.. presumably you actually need that much thruput :) |
| 17:23.38 | [hC] | i looked at ser and it looked like a config nightmare |
| 17:23.48 | justinu | ser is actually pretty simple |
| 17:23.56 | [hC] | then again I probably could have taken a better look. |
| 17:23.56 | freestyle_networ | were a new VoIP emerging Wholesale VoIP carrier here in Edmonton, Alberta ... we have 1 DS3 now with full SS7 links we actually have alot of * people peering with us for 800, DID's,etc |
| 17:24.06 | shodan | don't answer all at once now ;) |
| 17:24.09 | *** join/#asterisk ResidenteE (i=Angel200@201.236.213.203) |
| 17:24.21 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:24.21 | ast_freak | alvariux, interesting. Yes, I haven't had much time to play with realtime yet. Hoping to do so in the future. |
| 17:24.52 | freestyle_networ | justinu, SER Is simple ..somtimes looses its brains ..not suitable for tru 24/7 in my opnion ....try openser as well |
| 17:25.17 | justinu | freestyle_networ: yeah, i'm running openser |
| 17:25.55 | ResidenteE | hia all, this an offtopic but asterisk related question: when use a h323 or SIP gatekeeper all communications pass through server? |
| 17:26.24 | shodan | or ... what's the cheapest in wifi phones ? |
| 17:26.27 | [TK]D-Fender | [hC] : I own one. |
| 17:26.55 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:26.59 | [hC] | [TK]D-Fender played with it much? |
| 17:27.04 | *** join/#asterisk Samoied (n=Samoied@mx1.opens.com.br) |
| 17:27.21 | freestyle_networ | PakiPenguin, vendor is Metaswitch ..our upstream provider is YellowStream ..ahem .. i mean Allstream |
| 17:27.22 | rajiv|work | so if i plug a T1 line into a te110p can i have * handle the voice channels and linux ip tables or something handl the data lines ? |
| 17:27.33 | [hC] | Augh |
| 17:27.36 | [hC] | We use allstream too |
| 17:27.41 | PakiPenguin | i see :p |
| 17:27.42 | [hC] | They are a bit of a hassle to deal with. |
| 17:27.48 | freestyle_networ | hC, u have no idea |
| 17:27.49 | [TK]D-Fender | [hC] : A fair bit, whats your question? |
| 17:28.06 | freestyle_networ | hC, major beuracracy there ....they are laying off 25% of their staff actually, big shake ups |
| 17:28.07 | [hC] | They gave us great PRI rates, but for fucks sake, getting our billing sorted out, or number porting, anything, takes ages. |
| 17:28.25 | [hC] | i put in a number port request a month ago |
| 17:28.28 | [hC] | still havent heard back |
| 17:28.30 | [TK]D-Fender | [hC] : AllStream as a carrier? |
| 17:28.32 | freestyle_networ | hC, oh are you having fun with the order rejects for Number ports too???? :P |
| 17:28.56 | freestyle_networ | hC, yeah this realted to all the layoffs right now we think, no one their gives a fuck right now |
| 17:29.07 | shodan | am I transparent or something ?! jeez ! |
| 17:29.18 | justinu | canadian? |
| 17:29.18 | [TK]D-Fender | errr, freestyle_networ : that last one was for you... |
| 17:29.30 | justinu | shodan: no one knows the answer to your question, or feels like talking about it now |
| 17:29.34 | justinu | it's just the way IRC is |
| 17:29.40 | [hC] | freestyle_networ yeah actually my rep is moving from vancouver to edmonton, and he hasnt got back to me on SHIT, and now im dealing with another guy whos taking over the account who hasnt contacted me once. |
| 17:29.46 | freestyle_networ | yep they are a carrrier here in Canada, they used to be ALlstream Canada, MTS bought them out and started Allstream |
| 17:30.07 | justinu | i've been working with level3 on wholsale sip origination/termination |
| 17:30.09 | justinu | so far so good |
| 17:30.09 | [TK]D-Fender | freestyle_networ : I use them in Montreal here. I HATE them now.... |
| 17:30.10 | freestyle_networ | hC, who is your rep? |
| 17:30.11 | Strom_C | oh boy oh boy, Manitoba Telephone |
| 17:30.15 | [hC] | freestyle_networ: not to mention they promised us particular rates but never reflected it on our bills. dealing with getting that resolved has taken over 50 hours i'd bet |
| 17:30.30 | [hC] | freestyle_networ: the one that just moved is Gerry Peterson, the one we just received is Scott Lockey |
| 17:30.30 | freestyle_networ | hC, dude thats just telco, they all suck |
| 17:30.42 | [hC] | I thought allstream used to be at&tcanada |
| 17:30.43 | freestyle_networ | hC, yep we know SCott, hes a good shit |
| 17:30.49 | [TK]D-Fender | Allstream has &^%ed up EVERY SINGLE ORDER I've ever placed with them one way or another. |
| 17:30.51 | [hC] | Scott seems better than gerry |
| 17:30.56 | [hC] | but getting ahold of him is still fuckin hard. |
| 17:31.09 | shodan | justinu, I'd have thought that the price per fxs and per wifi phone would be something everyone knows here , kinda like the price of gas or milk |
| 17:31.12 | [hC] | just curious, since i cant get a single answer out of these guys... |
| 17:31.15 | freestyle_networ | hC, yep he is better |
| 17:31.21 | [TK]D-Fender | [hC] : They were previously AT&T Canada |
| 17:31.32 | justinu | shodon: some of us don't work in that "area" |
| 17:31.37 | *** join/#asterisk jasonwolfe0u812 (n=jasonwol@adsl-072-151-106-082.sip.asm.bellsouth.net) |
| 17:31.41 | [TK]D-Fender | shodan : What densities? |
| 17:31.42 | rajiv|work | shodan: voipsupply.com perhaps ? |
| 17:31.53 | [hC] | is it easier for me, for number porting, to contact someone OTHER than scott? Ive been contacting him, and then i dont even know how long its supposed to take, or how to check on it... Ive put in 3 requests so far, and none have completed, and my customers are getting suspicious. |
| 17:32.13 | freestyle_networ | hC, yeah hang tight, the shake ups are just happening this month ....they just canned the sales manager at Edmonton and their isnt anymore here ...were actually the 4th biggest allstream reseller in Canada and are trying to take over their edmonton base :D |
| 17:32.17 | justinu | hc: sounds like you need to be more of a pain in the ass |
| 17:32.26 | shodan | [TK]D-Fender, low , 5-10-20 fxs ports in the dirt cheap rough edge chinese made kind |
| 17:32.27 | jasonwolfe0u812 | anyone know if I will get answer supervision if I user an IAX server to terminate voip calls into a pstn so that my script will wait to fire |
| 17:32.31 | [hC] | justinu: i have been, trust me. |
| 17:32.41 | freestyle_networ | hC, screaming and yelling often doesnt do shit to telecom process |
| 17:32.52 | justinu | program one of your switches to spam all the reps with calls until they call you back :) |
| 17:33.02 | justinu | on the hour, every hour :) |
| 17:33.07 | [hC] | I just contacted their boss, bo, and expressed concern and threatened to look for another carrier. |
| 17:33.08 | freestyle_networ | hC, but out of all of them ALlstream is the most forward thinking here in Canada, they are still the only ones offering VoIP PRI's ..ie: e911 |
| 17:33.28 | shodan | rajiv|work, I'll take a look thanks |
| 17:33.28 | justinu | what's a VoIP PRI? |
| 17:33.30 | freestyle_networ | hC, we save your breath on that, they get that alot..trust me on this |
| 17:33.42 | freestyle_networ | jstinu, its a PRI with 911 ability |
| 17:33.47 | Beirdo | OMG |
| 17:33.48 | [hC] | yeah. i just have standard pri from them, in peer1 downtown vancouver. we just direct cross connect to them on the meetme floor |
| 17:33.48 | justinu | oh |
| 17:33.52 | freestyle_networ | or should is say e911 |
| 17:33.55 | justinu | how does that make it VoIP? |
| 17:34.00 | justinu | yeah, makes more sense :) |
| 17:34.04 | freestyle_networ | thats what they call it |
| 17:34.06 | Druken | it's not voip |
| 17:34.09 | justinu | weird |
| 17:34.21 | freestyle_networ | we actually do sell true VoIP PRI's, you peer to use for 24 channels |
| 17:34.22 | Beirdo | 9428m 26s of calls in December |
| 17:34.24 | Druken | it's a voip carriers pri.. that allows entry into the 911 system |
| 17:34.25 | Beirdo | JEEEZ |
| 17:34.26 | freestyle_networ | SIP only of course |
| 17:34.28 | [TK]D-Fender | shodan : The lowest I've seen is aroun $100USD / port once you hit 4 ports. (Clipcomm occasionally drops to 87.5/port) |
| 17:34.40 | justinu | druken: canada 911 only? |
| 17:34.57 | [hC] | freestyle_networ: so, about numer ports. is there a different department i can contact to get stuff done quicker? or how long is that junk normally supposed to take? I dont really care how long it does take as long as i can tell my customers something consistent. |
| 17:34.59 | freestyle_networ | yeah, Allstream is canada only ..and only major centers from what i hear ..ie: if you live in the sticks, u wont get e911 |
| 17:35.06 | justinu | ah |
| 17:35.07 | [hC] | if i tell them 2 weeks and it takes 5, i look like a dumbass. |
| 17:35.07 | [TK]D-Fender | [hC] : What was your question about SPA-941's? |
| 17:35.16 | Druken | justinu: you add address entries for your assigned did's |
| 17:35.19 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:35.29 | freestyle_networ | hC, u know u should talk with our Pres, he know how to menuever thru Allstream much better |
| 17:35.44 | [hC] | [TK]D-Fender: sorry, got wrapped up in this convo here... :) Two things.. first of all, is it possible to do speed dials on the line buttons like you can on the 7940's? and secondly, what does their 'shared line appearance' actually do? |
| 17:35.56 | shodan | [TK]D-Fender, yikes ! I was eyeballing the Cisco MC3810 since they are 5 fxs 1 fxo and they go for about 200$usd+ship on ebay |
| 17:36.33 | *** part/#asterisk ResidenteE (i=Angel200@201.236.213.203) |
| 17:36.38 | [TK]D-Fender | [hC] : Doesn't look like so far, and * doesn't support "Shared Apperances" yet. Thats like being able to register in 2 places at once. |
| 17:36.39 | *** join/#asterisk Mo (i=dark@g-unit.ca) |
| 17:36.40 | *** join/#asterisk gniretar_work (n=mark@152.160.35.1) |
| 17:36.47 | gniretar_work | hi everyone |
| 17:37.09 | shodan | I'm still kinda in shock that they're not in the 10/20$usd per fxs shipped for "dumb" modules considering how little it takes to make a fxs |
| 17:37.10 | [TK]D-Fender | shodan : could work, but I don't know its standards and there is the question of support... |
| 17:37.10 | freestyle_networ | Druken, yeah its Paul , who dat? |
| 17:37.21 | Druken | [TK]D-Fender: shared apperances would be an addition to asterisk |
| 17:37.30 | [hC] | [TK]D-Fender: I figured thats what it was, but they called it 'their technology' so i thought maybe they'd done something special. |
| 17:37.31 | Druken | freestyle_networ: James |
| 17:37.35 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc11a.dialup.mindspring.com) |
| 17:37.39 | *** join/#asterisk agx (n=agx@ip-37-53.sn1.eutelia.it) |
| 17:37.41 | gniretar_work | so i know how to route calls based on number called (exten=>) but how to i do it based on calling number. There is a person who keeps calling us and i'd rather have a message played when they call then ahve it ring my phone |
| 17:37.44 | [TK]D-Fender | Druken : A nice one at that given how many devices support it. |
| 17:38.01 | shodan | [TK]D-Fender, yeah, I'll do my homework first , worst case I'll just bite the bullet and build a bunch from scratch |
| 17:38.05 | Druken | [TK]D-Fender: agreed |
| 17:38.07 | gniretar_work | or better yet, i'd like to have then redirected to my fax machine |
| 17:38.12 | fugitivo | gniretar_work: look for "exgirlfriend" feature |
| 17:38.22 | [TK]D-Fender | gniretar_work : GotoIF based on CALLERID(num) |
| 17:38.25 | justinu | shodan: lol, this is telecom... take whatever it should cost, and multiply by 10! |
| 17:38.37 | agx | Hi |
| 17:38.39 | agx | I can register to my VoIP provider from home, but cannot register another PBX i've putted online into the office. Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k: NOTICE: chan_sip.c:7708 handle_request: Registration from '<sip:gallo-shphone@172.16.1.4>' failed for 'MY_PUBLIC_NATTED_IP' |
| 17:38.43 | [TK]D-Fender | justinu : Not since commodity SIP (mostly) |
| 17:38.52 | justinu | it's certainly helped things |
| 17:38.53 | freestyle_networ | anyone here know if the 2.6 is better for timming versus 2.4 kernel with ztdummy? we seem to hit a wall with conferecing at 250 channels with a 2.4 kernel and ztdummy |
| 17:38.56 | justinu | but we're not there yet |
| 17:39.04 | shodan | justinu, yup ! |
| 17:39.12 | *** join/#asterisk ravenpi (n=ravenpi@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 17:39.12 | fugitivo | freestyle_networ: it should |
| 17:39.16 | freestyle_networ | Druken, James ...wassup! |
| 17:39.31 | justinu | when I started working in telecom, a 16 span T1 card for the switches I was using was $35,000 |
| 17:39.35 | [hC] | 2.6 seems much better. |
| 17:39.42 | Druken | freestyle_networ: not much... how in hell are you getting 250 channels into a meetme? |
| 17:39.44 | freestyle_networ | figutivo, wondering how many extra channels could be squeezed out with that ...our bottle neck is timming, not CPU load |
| 17:40.00 | Druken | am i to assume the ds3 is fully active now? |
| 17:40.10 | freestyle_networ | Druken, not using meetme ;) ...actually meetme got up to 600 in one room ...crappy voice quality though :P |
| 17:40.15 | *** join/#asterisk cnet2 (n=jjohn@200.122.157.91) |
| 17:40.16 | fugitivo | freestyle_networ: like Druken said, how in the hell are you getting so many channels into meetme? :) |
| 17:40.21 | freestyle_networ | Druken, it is! |
| 17:40.26 | Druken | sweet! |
| 17:40.35 | *** join/#asterisk ELE_VV_MSN (n=bdcfl@201.29.156.32) |
| 17:40.43 | Druken | got the ranges now? or still for just edmonton and calgary? |
| 17:40.50 | ELE_VV_MSN | hi people |
| 17:40.58 | cnet2 | why is it that asterisk has a 7+sec delay from the moment i instruct a call and the moment the phone starts ringing |
| 17:41.03 | freestyle_networ | Druken, yes u will b ported ;) ..we have 905 and 416 up as well ..sorry no 750 ...always forget about Berry, Ontario ;) |
| 17:41.21 | Druken | barrie... |
| 17:41.21 | [TK]D-Fender | freestyle_networ : you work for a VoIP termination co? |
| 17:41.25 | Druken | and it's 705 :) |
| 17:41.30 | freestyle_networ | Druken, we have 604,780,403,905,416 right now |
| 17:41.37 | freestyle_networ | oops, sorry |
| 17:41.43 | justinu | how many area codes does canada have? |
| 17:41.48 | Druken | should have parts of 519 as well |
| 17:41.49 | [TK]D-Fender | justinu : quite a lot. |
| 17:41.50 | [hC] | alot. |
| 17:41.57 | justinu | probably about as many as california |
| 17:42.05 | [hC] | hahah |
| 17:42.05 | justinu | 40 or so? |
| 17:42.11 | [hC] | more than that, i think. |
| 17:42.34 | [hC] | maybe not.. im not sure. |
| 17:42.37 | justinu | back in the days, i remember when they had only one or 2 for each province |
| 17:42.42 | ELE_VV_MSN | anyone here have some sample about how to configure an FXS?? |
| 17:42.44 | justinu | it was easy to route |
| 17:42.51 | Druken | shit, toronto has 4.... |
| 17:43.00 | *** join/#asterisk ravenpi (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 17:43.11 | Druken | 416, 905, 647, 249 |
| 17:43.19 | [hC] | vancouver has 2, northern bc and the island have another |
| 17:43.35 | asteriskmonkey | whats this about toronto area codes? |
| 17:43.45 | justinu | i think LA has about 10 |
| 17:43.46 | *** join/#asterisk ravenpi_ (n=ravenpi@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 17:43.50 | justinu | in a 100 mile radius |
| 17:43.53 | [TK]D-Fender | Montreal (metropolitain area) has 2 so far, and is about to get a 3rd. |
| 17:44.09 | Druken | 514 and i forget the other... |
| 17:44.13 | [TK]D-Fender | 450 |
| 17:44.16 | Druken | :) |
| 17:44.17 | justinu | there was a big expansion in the middle 90s |
| 17:44.24 | [TK]D-Fender | And we're getting another this year |
| 17:44.31 | asteriskmonkey | is there anyone here with a pri that wants to setup a dundi network :) |
| 17:44.32 | [hC] | i just wish they took the fucking 16 or so rate centers out of vancouver |
| 17:44.36 | [hC] | and made it one. |
| 17:44.40 | Strom_C | los angeles has..... 213 323 310 818 805 562 714 949 626 909 951 424 |
| 17:44.41 | [hC] | number. porting. hell. |
| 17:44.48 | asteriskmonkey | is there any public dundi networks people can join? |
| 17:44.52 | justinu | 12 now, then :) |
| 17:45.03 | justinu | 951 and 424 are new to me |
| 17:45.10 | Druken | i'm waiting for the cellular numbers to become public |
| 17:45.10 | Strom_C | oh, i forgot 661 and 760 |
| 17:45.12 | Druken | hehe |
| 17:45.26 | Strom_C | 951 is a split from 909, 424 is the new 310 overlay code |
| 17:45.31 | justinu | joy |
| 17:45.39 | PakiPenguin | brb |
| 17:45.41 | ELE_VV_MSN | who have a sample code about how to use an FXS port?? |
| 17:46.01 | Druken | freestyle_networ: what do you mean i'll be ported? to the ds3? you guys getting rid of the pri ? |
| 17:46.02 | justinu | any of you guys use jabber? |
| 17:46.16 | *** join/#asterisk seele_ (n=seele@200.124.172.72) |
| 17:46.17 | Druken | use to |
| 17:46.45 | agallo | Did i done something wrong inside sip.conf ? I can register to my VoIP provider from home, but cannot register another PBX i've putted online into the office. NOTICE: chan_sip.c:7708 handle_request: Registration from '<sip:gallo-shphone@172.16.1.4>' failed for 'MY_PUBLIC_NATTED_IP' |
| 17:46.45 | justinu | jabber confernece rooms are so far ahead of IRC, it's sick |
| 17:46.50 | [TK]D-Fender | ELE_VV_MSN : What kind of interface? |
| 17:46.59 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 17:47.00 | gniretar_work | justinu: I use it all the time, GTalk :-D |
| 17:47.10 | justinu | googletalk sucks tho. |
| 17:47.12 | Mo | hrm, anyone got a few minutes to spare? |
| 17:47.18 | justinu | they don't allow s2s connections |
| 17:47.21 | justinu | bastages |
| 17:47.23 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 17:47.24 | [hC] | agallo: just a thought, but do you have "my_public_natted_ip" listed in host= in sip.conf? |
| 17:47.25 | gniretar_work | justinu: its new, it will catch up |
| 17:47.29 | seele_ | Hi, does anyone knows, why when i make a call from an extension to my own PBX. it makes some kind of a "loop", so that no more calls are recieved or made, even when hanging up |
| 17:47.44 | Dandan | hey all |
| 17:47.53 | Dandan | can't get parking calls announced on gxp-2000 |
| 17:47.55 | Dandan | any ideas? |
| 17:48.05 | Dandan | after i transfer to 700 it hangs up |
| 17:48.10 | gniretar_work | seele_: I'm an asterisk noob but i'd say u should look at what the debug console says and post it |
| 17:48.17 | agallo | [hC]: no in sip.conf i use host=dynamic |
| 17:49.54 | ELE_VV_MSN | \nick CPC-BR |
| 17:50.51 | gniretar_work | how do i stack conditions on gotoif? |
| 17:50.58 | gniretar_work | is it possable? |
| 17:51.57 | CPC-BR | anyone here speak portuguese??? |
| 17:52.24 | cnet2 | i speak spanish, i could probably help you |
| 17:52.42 | JunK-Y | gniretar_work: show application Gosub ? |
| 17:53.16 | [TK]D-Fender | gniretar_work : using the IF function, yes. |
| 17:53.55 | [TK]D-Fender | gniretar_work : although I find it easier to just execute multiple tests in sequence. |
| 17:54.02 | gniretar_work | k |
| 17:54.03 | [TK]D-Fender | gniretar_work : how complex? |
| 17:54.13 | gniretar_work | too complex |
| 17:54.18 | gniretar_work | i found a better way to do it |
| 17:54.27 | *** join/#asterisk implicit (n=implicit@200.12.227.205) |
| 17:54.33 | gniretar_work | thx tho. I relocated the statement so i only need 1 conidtion |
| 17:54.40 | justinu | word, implicit |
| 17:54.58 | agallo | this is my client config in sip.conf: [agallo] context=interni type=friend username=agallo secret=agallo regexten=201 host=dynamic canreinvite=no qualify=yes nat=yes ... this work if i'm inside LAN not if i'm connected from another office |
| 17:55.34 | *** join/#asterisk Kernel_Core (n=I@92.230.dial-up.xter.net) |
| 17:55.40 | Kernel_Core | hi all .. |
| 17:55.41 | Druken | agallo: what's in your global ? |
| 17:55.50 | Druken | and uhmm... use a pastebin |
| 17:55.53 | agallo | Druken: just a second |
| 17:56.39 | agallo | Druken: nat=yes externip=PBX_PUBLIC_IP localnet=192...ecc/255...eccc |
| 17:56.52 | Druken | lookie there.... |
| 17:57.04 | Druken | externip=PBX+PUBLIX_IP |
| 17:57.16 | Druken | how bout ya set that to your ip |
| 17:57.41 | Kernel_Core | I have 2 asterisk server , server A is behiend NAT , Server B has Public IP , both servers are connected through IAX (with register.... command in iax.conf) when I want to generate a call from B Server to A Server I get this error " Rejected connect attempt from 193.222.108.40, who was trying to reach 's@' |
| 17:57.58 | implicit | word |
| 17:58.13 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 17:58.16 | *** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com) |
| 17:58.29 | Strom_C | is there any way to run "zap show status" from the command line? |
| 17:58.30 | Druken | Kernel_Core: might wana tell it what context |
| 17:59.05 | JunK-Y | Strom_C: huh? from the CLI: sure, just type it :) |
| 17:59.05 | dalabera | asterisk -rx zap show status |
| 17:59.16 | iDunno | asterisk -rx "zap... dammit beaten to it. |
| 17:59.21 | CPC-BR | I have to make a call using an FXS port...but I don't an FXS board...I have just to signilize this calling...is it possible?? the asterisk server have just to show that the calling was made...have to generate an txt file or something like that |
| 17:59.21 | agallo | Druken: sorry i forget to mention that PBX receive data thru port forwarding; so nat=yes and externip=PUBLIC_IP_OF_FIREWALL |
| 17:59.40 | Kernel_Core | Druken: I defined in general segment of iax.conf "context =>loc110" |
| 17:59.44 | Kernel_Core | is it right ? |
| 18:00.04 | Druken | well, it's right... but it doesn't seem to be using it |
| 18:00.07 | *** join/#asterisk Brumle (n=brumle@brumle.com) |
| 18:00.18 | Mo | where can I get more information about asterisk's professional solutions? |
| 18:00.21 | Kernel_Core | [loc110] |
| 18:00.24 | Druken | agallo: ya still need to set the ip |
| 18:00.34 | JunK-Y | Mo: its called Business Edition. |
| 18:00.37 | Kernel_Core | exten => _XXXXX.,1,Dial(Zap/g1/${EXTEN}) |
| 18:00.37 | Kernel_Core | exten => _XXXXX.,2,Congestion |
| 18:00.38 | agallo | Druken: what do you mean? :))) |
| 18:00.54 | JunK-Y | just see their website. |
| 18:00.56 | Druken | what is your public ip ? |
| 18:01.01 | Mo | thanks. |
| 18:01.21 | ManxPower | *SCREAM* I need to do Polycom BLF today. |
| 18:01.24 | Kernel_Core | Druken: when I directly connect to "A" Server I can place call with this context .... |
| 18:01.25 | ManxPower | Evil users. |
| 18:01.30 | agallo | Druken: FW is on 80.0.0.0 it port forwards UDP ports to 172.0.0.0 net |
| 18:02.10 | *** join/#asterisk SpaceBass (n=SP@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 18:02.15 | SpaceBass | howdy |
| 18:02.15 | *** join/#asterisk umay (n=chris@65-37-2-236.nrp2.roc.ny.frontiernet.net) |
| 18:02.24 | dalabera | guys quick question, I have installed a new server for *, don't want to use MOH because the problems it carries, can I use a wav to playback continously for incoming calls to queue? |
| 18:02.34 | Druken | agallo: so? that's a forward for incoming packets, asterisk needs to know what ip it's sending from... |
| 18:02.36 | SpaceBass | anyone know who the user should be for files in my /tftpboot dir (asterisk@home) |
| 18:02.37 | Kernel_Core | Druken: it seems something is wrong , asterisk which is running in "A" Server , can't get incomming call and fw it to one of it's ZAP interfaces.... |
| 18:02.39 | fugitivo | dalabera: what problems? |
| 18:03.10 | iCEBrkr | MOH has problems? |
| 18:03.11 | agallo | Druken: indeeed :))) externip=FirewallIP_on_80.0.0.0_network |
| 18:03.35 | Druken | remember it must be the PUBLIC address |
| 18:03.48 | agallo | Druken: i'm saying :) it is a public address |
| 18:03.51 | Beirdo | iCEBrkr: there are potential copyright issues with MOH usage, but I don't know if that's what he means :) |
| 18:03.58 | fugitivo | iCEBrkr: that's new for me |
| 18:04.00 | dalabera | let's say I have 15 calls on hold, I've seems that * start to slow down, the processor goes up, and besides new problems arises |
| 18:04.06 | iCEBrkr | Beirdo: Oh please... |
| 18:04.07 | *** join/#asterisk NDT (n=me@cpe-24-195-216-41.nycap.res.rr.com) |
| 18:04.15 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 18:04.19 | agallo | Druken: i get this error when registering only from remote (outside lan) NOTICE: chan_sip.c:7708 handle_request: Registration from '<sip:gallo-shphone@172.16.1.4>' failed for 'MY_PUBLIC_NATTED_IP' |
| 18:04.21 | dalabera | when using mpg123 |
| 18:04.29 | xachen | use madplay :) |
| 18:04.31 | xachen | its so much better |
| 18:04.37 | Beirdo | iCEBrkr: it depends on how you choose to use it, but there are potential issues with broadcasting, blah blah blah |
| 18:04.40 | iCEBrkr | dalabera: There's some sort of native MoH now with doesn't use mpg123 |
| 18:04.40 | fugitivo | dalabera: use native moh so you don't need extra programs |
| 18:04.44 | Beirdo | we tend to ignore it, but it's there |
| 18:04.53 | [TK]D-Fender | CPC-BR : What is your FXS using? (hardware) |
| 18:04.55 | xachen | and you can get royalty free mp3 music for MOH |
| 18:05.06 | Beirdo | yes |
| 18:05.08 | dalabera | Native Moh, let me check that out.... :-) |
| 18:05.11 | iCEBrkr | Beirdo: Whatever, I'll whip out the ol' keyboard and compose my own MoH. |
| 18:05.18 | Beirdo | hehe |
| 18:05.26 | iCEBrkr | Beirdo: Then I'll sue myself for using copyrighted material. |
| 18:05.27 | fugitivo | dalabera: 1.2.x addons only |
| 18:05.35 | Beirdo | as long as it's not like Ross from Friends :) |
| 18:05.47 | dalabera | oh |
| 18:05.53 | dalabera | cool thanks |
| 18:05.54 | [TK]D-Fender | iCEBrkr : I had a few classical guitar bits of mine on my company's MOH till I found some freebies :) |
| 18:06.07 | dalabera | thanks god I got it installed!! |
| 18:06.37 | iCEBrkr | There's actually some good S3Ms and MOD's out there that aren't too bad.. Believe it or not they don't sound like Nintendo 64 crap |
| 18:07.39 | fugitivo | skaven |
| 18:07.46 | fugitivo | anyone know skaven? |
| 18:07.49 | iCEBrkr | Hell, fire up Acid DJ and loop some sound bites. :) |
| 18:07.50 | CPC-BR | [[TK]D-Fender] no one. I really need an FXS hardware? |
| 18:07.56 | *** join/#asterisk _DAW (n=bob@adsl-156-94-42.msy.bellsouth.net) |
| 18:08.19 | fugitivo | the old future crew scene group :) |
| 18:08.27 | iCEBrkr | fugitivo: Rock'on!! Purple Motion |
| 18:08.40 | iCEBrkr | Purple Motion STILL composes some good stuff |
| 18:08.47 | fugitivo | skaven too! |
| 18:08.52 | [hC] | iCEBrkr: you suck! |
| 18:08.54 | CPC-BR | [[TK]D-Fender] I just want to simulate this calling |
| 18:08.56 | iCEBrkr | [hC]: You mom. |
| 18:08.57 | iCEBrkr | err |
| 18:08.58 | iCEBrkr | Your |
| 18:09.04 | [hC] | Durr.. |
| 18:09.07 | [TK]D-Fender | CPC-BR : well you said you had an FXS problem. That REQUIRES hardware for it to be a problem :) |
| 18:09.13 | iCEBrkr | [hC]: :P |
| 18:09.21 | CPC-BR | [[TK]D-Fender] :) |
| 18:09.42 | iCEBrkr | [hC]: How aboot ya fetch me some Molsen, eh? |
| 18:09.49 | *** join/#asterisk fulco (i=fulco@d-ip-129-15-215-141.wireless.ou.edu) |
| 18:09.53 | _DAW | Hello |
| 18:10.03 | _DAW | Could somehone here help me with the proper use of the VMCOUNT() function? |
| 18:10.18 | [TK]D-Fender | CPC-BR : Just make an extension that leads to where your calls will land. |
| 18:11.52 | CPC-BR | [[TK]D-Fender] I want that the * server show something to prove that the calling was make...did u understand? |
| 18:12.05 | *** join/#asterisk chapeaurouge (n=chapeaur@85.201.81.201) |
| 18:13.17 | [TK]D-Fender | You are trying to simulate an incoming call on an interface you don't have. just set it up so you can dial the exten that is your incoming context and thats it. |
| 18:15.05 | SpaceBass | any hot tops for a nasty phone fuck? |
| 18:15.25 | [TK]D-Fender | ..... |
| 18:15.27 | justinu | nasty phone fuck? lol |
| 18:15.35 | CPC-BR | [[TK]D-Fender] man my english is not very good :) ... but I'm not sure that I'm trying to simulate an incoming call. |
| 18:15.42 | [TK]D-Fender | Get a Grandstream... you'll really be fucked then! |
| 18:16.08 | SpaceBass | LOL |
| 18:16.23 | SpaceBass | and that folks is why you don't walk away from you keybard |
| 18:16.23 | Beirdo | OMG, I spent 18h on the phone already this month?! |
| 18:16.24 | CPC-BR | [[TK]D-Fender] i want to do the following situation - PC1 -> *Server -> PSTN |
| 18:16.56 | [TK]D-Fender | CPC-BR : do you have PC -> * working right? (audio both ways, dial plan setup) |
| 18:16.58 | SpaceBass | i leave for 15 minutes and my so call friends have pulled up p0rn all over my computer and done lord knows what else |
| 18:17.09 | Beirdo | sent pr0n to the boss? |
| 18:17.26 | Beirdo | this is why ya lock the kybard |
| 18:17.29 | Beirdo | :) |
| 18:17.34 | SpaceBass | I'm usually religious about it... |
| 18:17.38 | SpaceBass | payback will be a bitch |
| 18:17.44 | CPC-BR | [[TK]D-Fender] yeap |
| 18:18.56 | CPC-BR | [[TK]D-Fender] i already configure the * server to link 2 PCs at the same lan. PC1 -> *server -> PC2 (this situations is workin perfectly) |
| 18:19.09 | hackeron | [TK]D-Fender: not sure if you saw my message above, but I got it working! - WOOPEE!!!! - ManxPower helped me out by suggesting localnet= which made it spring into life. |
| 18:19.27 | ManxPower | hackeron, its all documented in the Wiki |
| 18:19.28 | hackeron | [TK]D-Fender: thanks for all your help, its working lovely :) |
| 18:19.33 | [TK]D-Fender | CPC-BR : ok, so you need help setting up a land line? |
| 18:19.41 | [TK]D-Fender | hackeron : Good to hear! |
| 18:19.55 | [TK]D-Fender | hackeron : Yeah you kinda need that :) |
| 18:20.00 | hackeron | ManxPower: the wiki is quite big, but you're right, I should read the documentation better. Thanks for all your help! |
| 18:20.10 | asterisk99 | anyone hear of a problem where the "Comedian" of "Comedian Mail" announcement being replaced with a "beep"???? |
| 18:20.18 | CPC-BR | [[TK]D-Fender] yeap |
| 18:21.09 | asterisk99 | (English majors can refrain from commenting on my improper use of the Queen's English) |
| 18:22.09 | [TK]D-Fender | CPC-BR : What are you using to access the PSTN? |
| 18:22.20 | *** join/#asterisk PakiPenguin_ (i=uppal@linuxpakistan/admin/pakipenguin) |
| 18:22.35 | *** join/#asterisk Cyberchen (n=cyberche@access.comba.ch) |
| 18:22.44 | devel | anybody here using an allied telesyn branded ATA against asterisk SIP? |
| 18:24.02 | *** join/#asterisk E|nyPRI_ (n=les@iphost-64-56-141-113.wpg.wiband.net) |
| 18:24.05 | *** join/#asterisk astneb (n=no@113-20-17.adsl.cust.tie.cl) |
| 18:24.08 | E|nyPRI_ | H. |
| 18:24.09 | ManxPower | Any parking gurus here? |
| 18:24.10 | E|nyPRI_ | Hi. even. |
| 18:24.24 | astneb | hi, can someone help me with a digium card issue? |
| 18:24.28 | E|nyPRI_ | Does anyone know how to set the hangup code=34 on the nufone h323 driver? |
| 18:24.35 | *** part/#asterisk agallo (n=agx@ip-37-53.sn1.eutelia.it) |
| 18:24.58 | Druken | ManxPower: get valet parking :) |
| 18:25.06 | CPC-BR | [[TK]D-Fender] i dont want to access the PSTN because I dont have an FXS hardware. I just want the * server generate some information to prove that its is possible..but without an FXS hardware...i dont want to complete de calling..did u understand? |
| 18:25.21 | astneb | the problem is that i have a tdm400p and when i dial out in the CDR shows as answered all the time, even if the call didn't went through |
| 18:25.29 | ManxPower | Druken, I suppose I could spend a day getting it into my build process. |
| 18:25.37 | *** join/#asterisk NDT (n=me@cpe-24-195-216-41.nycap.res.rr.com) |
| 18:25.40 | ManxPower | Ahrimanes, so there are no parking gurus here. |
| 18:26.45 | rajiv|work | CPC-BR: it is possible bc all of us with TDM cards do it. but it would be hard to get * to place a PSTN call without the hardware |
| 18:26.55 | [TK]D-Fender | CPC-BR : just create a dial pattern and playback the number to test it. |
| 18:26.56 | Druken | ManxPower: what is messing you up ? |
| 18:26.57 | *** join/#asterisk kippi1 (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
| 18:26.59 | kippi1 | hey |
| 18:27.40 | [TK]D-Fender | CPC-BR : like "exten => _9x.,1,Answer", "exten _9x.,2,Playdigits(${EXTEN:1})" |
| 18:27.45 | kippi1 | can someone have a look at this please http://forums.digium.com/viewtopic.php?t=3506&highlight= |
| 18:27.48 | *** join/#asterisk twisty7867 (n=twisty78@adsl-gte-la-216-86-203-111.mminternet.com) |
| 18:27.50 | *** part/#asterisk SpaceBass (n=SP@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 18:28.17 | astneb | anyone? |
| 18:28.23 | ManxPower | Druken, when a parked call times out in 1.2 it goes to: |
| 18:28.24 | ManxPower | <PROTECTED> |
| 18:28.24 | ManxPower | <PROTECTED> |
| 18:28.24 | ManxPower | <PROTECTED> |
| 18:28.50 | ManxPower | I sort of assumed I would have a little better control over this. In 1.0.x it always timed out to exten s |
| 18:29.04 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:29.25 | Druken | when it times out, it should go back to the device that parked it |
| 18:29.31 | twisted[asteria] | yup |
| 18:29.36 | CPC-BR | [[TK]D-Fender] ok man...i'll try to do that..thx |
| 18:30.07 | [TK]D-Fender | CPC-BR : Trust us, thousands of people use * with PSTN all over the world. It works. |
| 18:30.09 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 18:30.43 | ManxPower | I think I just crashed Asterisk |
| 18:31.00 | CPC-BR | [[TK]D-Fender] I know, but they do not use without an FXS hardware :) |
| 18:31.03 | [TK]D-Fender | ManxPower : suckcess! |
| 18:31.34 | [TK]D-Fender | CPC-BR : Some use a VoIP provider to lead to the PSTN. Either way you're just "faking" it. |
| 18:32.20 | CPC-BR | VoIP provider?? what do u mean with it? |
| 18:33.00 | [TK]D-Fender | CPC-BR : a company that will sell you a phone number and route calls to/from your server using it over SIP/IAX2. |
| 18:33.13 | kippi1 | can anyone help with my outdialing error? |
| 18:33.45 | xachen | How do I get native MOH to work? |
| 18:33.50 | xachen | i have 1.2.0 |
| 18:34.09 | astneb | the problem is that i have a tdm400p and when i dial out in the CDR shows as answered all the time, even if the call didn't went through |
| 18:35.04 | CPC-BR | [[TK]D-Fender] could u tell me some companies name? |
| 18:35.13 | ManxPower | ~mailinglist |
| 18:35.16 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 18:35.29 | *** part/#asterisk twisty7867 (n=twisty78@adsl-gte-la-216-86-203-111.mminternet.com) |
| 18:36.10 | [TK]D-Fender | CPC-BR : Check the WIKI, there are dozens of them... |
| 18:36.26 | CPC-BR | [[TK]D-Fender] ok |
| 18:36.48 | CPC-BR | [[TK]D-Fender] thx very much :) |
| 18:36.55 | Druken | astneb: live with it... welcome to analog, only way to even remotely fix it is callprogress=yes in zapata.conf and i don't reccomend it |
| 18:37.37 | Druken | or do what i do, and only use the analog lines for incoming, send everything out a voip carrier |
| 18:38.09 | justinu | yeah, i guess there's no way to do real answer supervision on a 2 wire line |
| 18:39.03 | astneb | Druken: whats the problem with callprogress=yes? |
| 18:39.10 | justinu | unreliable, number 1 |
| 18:39.15 | *** join/#asterisk darkskiez (n=darkskie@bb-195-172-50-11.ukonline.co.uk) |
| 18:39.20 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 18:39.28 | justinu | probably requires a lot more CPU resources |
| 18:39.34 | tainted_ | any wiring gurus here? |
| 18:39.52 | tainted_ | i've got a client who wants to get rid of her landline completely at her HOME |
| 18:40.01 | justinu | easy |
| 18:40.03 | tainted_ | she's got around 10 phones total around the house |
| 18:40.10 | Druken | astneb: doesn't work worth a shit... sometimes it continues to ring even after they have answered |
| 18:40.12 | tainted_ | and chose to go with an ATA |
| 18:40.24 | ManxPower | Has anyone gotten RPID stuff working with Polycom |
| 18:40.29 | justinu | ManxPower: yes |
| 18:40.32 | Druken | tainted_: simple... plug an ata into the demarc |
| 18:40.36 | tainted_ | should i just drop the ATA into the phonebox outside the house? |
| 18:40.44 | astneb | Drunken: thx |
| 18:40.51 | justinu | tainted_: more than likely, yes, but it should be protected from the elements |
| 18:41.04 | xachen | mushc beter :) |
| 18:41.10 | xachen | got Native MOH working with .raw files |
| 18:41.12 | [TK]D-Fender | tainted_ : you should be able to trace to the demarc point inside the house and cut off the telco there and insert the ATA |
| 18:41.12 | justinu | tainted_: make sure you disconnect the CO pair from the house wiring too |
| 18:41.12 | tainted_ | it's hard to get electricity and ethernet out to the box |
| 18:41.14 | Druken | tainted_: make a new box inside the basement |
| 18:41.21 | kippi1 | i am really pulling my hair out here!! i am so close to getting a working PBX but so far at the same time!! can anyone shed any light on my problem? |
| 18:41.30 | tainted_ | i guess i could create a PoE breakout box.. but is there a better way? |
| 18:41.33 | ManxPower | justinu, I endabled sendrpid and trustrpid in sip.conf [general]. anything else I need to do. It's not well documented. |
| 18:41.45 | justinu | ManxPower: what exactly do you want to accomplish? |
| 18:42.12 | ManxPower | justinu, When I call Robert Dobbs's SIP phone on my Asterisk server from another SIP phone, I want the To: Robert Dobbs |
| 18:42.14 | tainted_ | [TK]D-Fender is the demarc point the phonebox outside the house? |
| 18:42.25 | justinu | ManxPower: ahhh... i had to write a patch to do it. |
| 18:42.32 | justinu | asterisk won't do that right now. |
| 18:42.33 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 18:42.41 | kuku5 | How do I reload sip.conf without restarting * ? |
| 18:42.45 | ManxPower | justinu, Just what DOES asterisk do with RPID? |
| 18:42.57 | ManxPower | kuku5, "reload" |
| 18:42.57 | tainted_ | [TK]D-Fender if the PSTN is not providing service, do they still send voltage down the copper? |
| 18:43.02 | ManxPower | or reload chan_sip.so |
| 18:43.09 | justinu | ManxPower: the reason you don't get To:Robert Dobbs, is because * needs to send RPID to the CALLING phone with the 183 progress or 180 ringing |
| 18:43.16 | justinu | * doesn't do that in stock form |
| 18:43.24 | tainted_ | [TK]D-Fender b/c i was thinking of just plugging the ATA into a spare phone jack and making that the new demarc point |
| 18:43.26 | ManxPower | justinu, So what GOES Asterisk's RPID stuff so? |
| 18:43.27 | justinu | all asterisk does with RPID is puts it in the invite |
| 18:43.33 | [TK]D-Fender | tainted_ : Don't risk it.... it'll FRY something when you least expect it. |
| 18:43.37 | justinu | so when you call out, it shows who made the call |
| 18:43.39 | ManxPower | justinu, what is the effect of that? |
| 18:43.48 | justinu | RPID is sip's caller id |
| 18:43.51 | [TK]D-Fender | tainted_ : you can do that, just make SURE the outside is cut off. |
| 18:44.02 | justinu | ManxPower: what you're talking about is 'connected party id' |
| 18:44.17 | tainted_ | or i could disable the voltage pins from the ATA |
| 18:44.22 | Druken | tainted_: make sure the ata can support that many phones too... :) |
| 18:44.39 | justinu | RPID in outbound invite == calling party id |
| 18:44.39 | ManxPower | justinu, Ah! So there's really no functional difference, since Asterisk will accept the callerid info from the calling phone anyway. |
| 18:44.45 | tainted_ | Druken why not? it's just one single line |
| 18:44.49 | [TK]D-Fender | tainted_ : so much easier to just find where it comes in. usually they set up a terminal which is easy to disconnect. |
| 18:44.50 | justinu | RPID in 180/182 == connected party id |
| 18:44.50 | kippi1 | is there away i can make sure there isn't a fault with my lines using asterisk? and to make sure everything is configed |
| 18:44.53 | tainted_ | sharing one dialtone |
| 18:44.53 | rob0 | J. Robert Dobbs? J.R. "Bob" Dobbs? |
| 18:45.03 | Druken | tainted_: 1 line... but 10 phones... |
| 18:45.08 | ManxPower | justinu, so, as far as I can tell rpid does nothing useful. |
| 18:45.18 | [TK]D-Fender | tainted_ : Oh, and watch out for REN's.... |
| 18:45.20 | tainted_ | Druken do u mean the power draw? |
| 18:45.21 | ManxPower | rob0, Slack Rules! Hail Eris! |
| 18:45.25 | justinu | ManxPower: the problem arises when you get connected to someone you didn't call by way of call forwding or order dialplan issues |
| 18:45.25 | rob0 | :) |
| 18:45.32 | [TK]D-Fender | Eris? |
| 18:45.39 | Druken | tainted_: yeah |
| 18:45.40 | tainted_ | [TK]D-Fender how do u get the cat5 & power out to the demarc tho? |
| 18:45.42 | justinu | ManxPower: but at this stage, no, RPID does nothing for you |
| 18:45.47 | ManxPower | justinu, I understand why it's not working the way I thought it would. |
| 18:45.48 | rob0 | Today is Pungenday, the 3rd day of Chaos in the YOLD 3172 |
| 18:45.52 | [TK]D-Fender | tainted_ : A really big drill :D |
| 18:45.56 | tainted_ | [TK]D-Fender you just draw cable out? |
| 18:45.57 | E|nyPRI_ | Does anyone know how to set the hangup code=34 on the nufone h323 driver? |
| 18:45.58 | tainted_ | LOL |
| 18:45.58 | ManxPower | I just can't see ANY usefullness for it. |
| 18:45.59 | justinu | asterisk RPID support is basically for dealing with VoIP termination providers. |
| 18:46.04 | tainted_ | k |
| 18:46.04 | Druken | tainted_: i think the linksys will do upto 8 phones... |
| 18:46.06 | justinu | who need it. |
| 18:46.14 | [TK]D-Fender | I mean demarc within the house. its on the INSIDE before splitting off. |
| 18:46.19 | *** join/#asterisk pegger (n=Peg@pool-68-163-192-85.bos.east.verizon.net) |
| 18:46.19 | ManxPower | justinu, thanks |
| 18:46.27 | [TK]D-Fender | and yeah just draw cable. |
| 18:46.27 | justinu | but I wrote a patch that makes * send RPID in 183/180 ringing so you can see who you're connected with. |
| 18:46.30 | ManxPower | justinu, can you send my your calling party id patch? |
| 18:46.35 | tainted_ | Druken i've got a grandstream 488 |
| 18:46.45 | ManxPower | I'll add it to my Asterisk+BTEL stuff. |
| 18:46.50 | justinu | ManxPower: yeah, i'll have to diff it |
| 18:46.56 | [TK]D-Fender | tainted_ : ICK! Grandsuck! |
| 18:47.00 | tainted_ | hmm.. might need to outsouce this electrical stuff |
| 18:47.14 | [TK]D-Fender | And if you diconnect the demarc you can plug your ATA wherever else you feel like actually.... |
| 18:47.18 | ManxPower | justinu, you didn't put in the bug tracker or send it off to OpenPBX did you? |
| 18:47.22 | Druken | tainted_: dunno about that one..., but think about it.. the ata has to RING all those damn phones at the same time |
| 18:47.23 | justinu | ManxPower: it isn't very heavily tested yet |
| 18:47.28 | dalabera | About using Native MOH, do I have to convert my mp3 to raw format for best performance? |
| 18:47.33 | justinu | ManxPower: not yet, no one seemed interested, and I got sick of talking about it :) |
| 18:47.40 | justinu | on the digium or openpbx side |
| 18:47.42 | tainted_ | Druken good point |
| 18:47.56 | tainted_ | but most of the phones have power supplies as well |
| 18:48.07 | tainted_ | the ATA just has to send voltage down the line |
| 18:48.15 | xachen | what is the diff between Asterisk and OpenPBX? |
| 18:48.23 | justinu | forks of the same project |
| 18:48.39 | justinu | as to the actuall differences, i'm not even sure yet |
| 18:48.45 | ManxPower | xachen, That is about the same as asking "What's the difference between the USA and the Soviet Union" -- in 1963 |
| 18:48.48 | justinu | openpbx has an autoconf based build system |
| 18:49.23 | tainted_ | Druken is there a device that could regulate the power draw if the ATA doesn't support 10 phones? |
| 18:49.24 | xachen | But OpenPBX is prtetty much Asterisk isn't it? |
| 18:49.28 | xbmodder_lappy | I am hungry |
| 18:49.32 | xachen | <PROTECTED> |
| 18:49.44 | justinu | tainted_: no, i think you just need a heavier duty ATA |
| 18:49.52 | pegger | I am having issues with my iax2, I know it runs on udp 4569 but when i tcpdump udp all I get is ntp stuff no aix stuff, and I am even do iax2 reload and nothing shows up, what is going on, I am using SVN-branch-1.2-r7709M |
| 18:50.10 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 18:50.10 | *** join/#asterisk anonymouz666 (n=anonymou@gw.ideavalley.com.br) |
| 18:50.28 | anonymouz666 | echo "SOFT HANGUP $agi_channel" |
| 18:50.29 | Druken | tainted_: perhaps a booster? or a small amp? hehe goto your local radio shack |
| 18:50.31 | anonymouz666 | is this correct? |
| 18:50.32 | tainted_ | justinu what's a heavier ATA |
| 18:50.37 | anonymouz666 | I just can't hangup the channel |
| 18:50.48 | anonymouz666 | from AGI |
| 18:51.08 | pegger | any ideas? |
| 18:51.48 | justinu | tainted: i'm not sure what the max REN is on them, have you looked into that? |
| 18:51.48 | rob0 | why did they fork openpbx? |
| 18:51.57 | anonymouz666 | anyone in here use soft hangup in AGI scripts? |
| 18:52.03 | justinu | rob0: tired of digium not paying attention to them, i think |
| 18:52.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:52.55 | anonymouz666 | I can hangup the channel from CLI, but I can't from AGI |
| 18:53.04 | anonymouz666 | echo "SOFT HANGUP $agi_channel" |
| 18:53.08 | anonymouz666 | I am using this |
| 18:53.16 | anonymouz666 | :( |
| 18:53.17 | rob0 | the site is NOT impressive. |
| 18:53.48 | justinu | there's a lot of sharp guys working on opbx |
| 18:53.52 | ManxPower | I may have to disable call parking in Asterisk |
| 18:54.00 | justinu | i wouldn't write it off so quickly. |
| 18:54.39 | ManxPower | There seems to no way to control where a parked call times out to. |
| 18:54.44 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 18:55.27 | rob0 | I'm not writing it off, but I'm not ready to switch, either. :) |
| 18:55.27 | kuku5 | How do I reload music on hold file? |
| 18:55.42 | justinu | rob0: understandable, but keep your eye on it :) |
| 18:55.43 | Druken | ManxPower: did you try a timeout inside the parkedcalls context? |
| 18:56.30 | ManxPower | Druken, nope. Normally that's configured in features.conf |
| 18:56.43 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 18:56.53 | Druken | worth a shot no? |
| 18:57.38 | *** join/#asterisk j2 (n=j2@207.181.0.86) |
| 18:57.49 | *** part/#asterisk j2 (n=j2@207.181.0.86) |
| 18:57.50 | dalabera | Quick Question About using Native MOH, do I have to convert my mp3 to raw format for best performance? |
| 18:58.07 | Druken | sox? |
| 18:58.21 | xachen | erm its best :) |
| 18:58.32 | xachen | otherwise your running processes to convert from mp3 to raw anyways |
| 18:58.38 | xachen | its just so much easier to convert to raw in the first start |
| 18:58.50 | xachen | http://www.orderlyq.com/asteriskqueues.html#moh |
| 18:58.55 | xachen | teachs you there |
| 19:00.02 | dalabera | thank you, man, there should be thousand of guys like you..... |
| 19:00.42 | Druken | there are.... |
| 19:00.51 | Druken | just most of them are also assholes... |
| 19:01.00 | justinu | lol |
| 19:01.06 | dalabera | wait... don't be nasty... |
| 19:01.11 | justinu | <- definite asshole |
| 19:01.17 | xachen | file called me an asshole the first time he saw me :P |
| 19:01.41 | Druken | can't ya feel the love? |
| 19:01.44 | xachen | he didn't notice my IRC client crapped out and thought I was being rude for askinga question on join haha |
| 19:01.54 | xachen | when I was ther 30 seconds before ^_^ |
| 19:04.24 | hackeron | I'm trying to set up meetme but I cant find any information about how to install the thing. Changelog shows its included with asterisk but when I try to use it I see Jan 3 09:01:12 WARNING[11819]: pbx.c:1690 pbx_extension_helper: No application 'meetme' for extension - any ideas? |
| 19:04.46 | file | xachen: did not! |
| 19:04.54 | file | hackeron: do you have a zaptel timing source? |
| 19:06.15 | hackeron | file: no, http://www.voip-info.org/wiki/index.php?page=Asterisk%20timer says I dont need one |
| 19:06.39 | file | where does it say that? |
| 19:07.03 | file | it says you need timing for two things: meetme and IAX2 trunking |
| 19:07.11 | file | it also says if you don't have zaptel hardware, you have 3 options |
| 19:07.24 | ManxPower | hackeron, you have to have zaptel installed when you build Asterisk. then listen to what file says |
| 19:08.20 | hackeron | ManxPower: it says I have 3 options, one of them is zaptel, I dont have zaptel |
| 19:08.43 | [TK]D-Fender | hackeron : You need to compile Zaptel to get ZTDummy even if you aren't planning on using Zaptel hardware. |
| 19:08.57 | ManxPower | hackeron, you have to have the zaptel DRIVERS installed, even if you don't have the hardware |
| 19:09.04 | [TK]D-Fender | Compile ZAptel, then recompile * |
| 19:09.16 | Druken | ztdummy == Zap Tel Dummy |
| 19:09.39 | hackeron | [TK]D-Fender: ManxPower: oh? -- hmm, ok. |
| 19:10.31 | [TK]D-Fender | YES |
| 19:11.29 | file | DJ Doboy! |
| 19:15.16 | *** join/#asterisk rva (n=Miranda@200.206.141.250) |
| 19:15.37 | NDT | is zttool realtime? Or only shows you status at the moment you run it? |
| 19:16.00 | rva | hi guys...does anyone know a reliable termination provider that offers unlimited call plans? |
| 19:16.29 | pegger | I am having issues with my iax2, I know it runs on udp 4569 but when i tcpdump udp all I get is ntp stuff no aix stuff, and I am even do iax2 reload and nothing shows up, what is going on, I am using SVN-branch-1.2-r7709M |
| 19:16.33 | wunderkin | rva, heh.. no one |
| 19:16.37 | mistral | i though most places are unlimited (other than the fact they charge you) |
| 19:16.42 | brad_mssw | rva: no, not 'unlimited' |
| 19:16.53 | brad_mssw | rva: 'unlimited' is usually a scam anyhow |
| 19:16.55 | kippi1 | whats the uk protocols for ISDN |
| 19:16.58 | Druken | termination to where? |
| 19:17.08 | rva | Druken: brazil |
| 19:17.11 | *** join/#asterisk woodchuck (n=woodchuc@S0106000000da2a3d.ok.shawcable.net) |
| 19:17.20 | Druken | newp, don't know of one |
| 19:18.00 | rva | there is this globaltelevoip.com that offers unlimited calls...it works...i used it |
| 19:18.04 | rva | but it is not that reliable |
| 19:18.10 | rva | sometimes the calls are not finished |
| 19:18.28 | hackeron | ManxPower: [TK]D-Fender: argh, I compiled it, but it says Invalid module format when I try to insmod :( |
| 19:19.10 | [TK]D-Fender | hackeron : Hmmm, maybe someone else here could help you with that one. It should work "out of the box" if you've got the std devel stuff |
| 19:19.33 | hackeron | [TK]D-Fender: well, its a kernel module, right? |
| 19:19.45 | hackeron | and I compiled 1.2.1 |
| 19:20.00 | *** join/#asterisk tsume (n=tsume@72.21.54.44) |
| 19:20.34 | tsume | is there a way to have my users available in the calling pool automatically without loggin in? or a way to automatically have asterisk log certain users in on boot? |
| 19:22.36 | *** join/#asterisk lnostdal (n=Lars@193.217.174.50) |
| 19:24.32 | [TK]D-Fender | hackeron : yes, but it should compile and install by itself |
| 19:25.02 | tsume | well nevermind. I'll just dial the phones from the server :) |
| 19:25.06 | *** part/#asterisk tsume (n=tsume@72.21.54.44) |
| 19:29.18 | *** join/#asterisk lorinc (n=ang@caracas-2941.adsl.interware.hu) |
| 19:34.55 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 19:39.56 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
| 19:40.10 | *** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net) |
| 19:40.25 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:40.30 | azzie | is there a way to debug DTMF events in * ? |
| 19:41.06 | azzie | on SIP channel |
| 19:43.16 | xachen | Does anybdoy happen to knowof a consol player (so I can link it into Asterisk) that supports MMS? |
| 19:43.49 | pegger | what is a consol player? |
| 19:43.58 | xachen | a CLI player rather :P |
| 19:44.15 | fugitivo | mplayer |
| 19:44.37 | xachen | thx :) |
| 19:50.13 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 19:50.41 | asterisk99 | anyone experience any sound problmes with IAXy (soft/hardphones)??? |
| 19:51.10 | Druken | has anyone else gotten some god damn outgoing ivr based questionare about a presence in iraq? |
| 19:51.28 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
| 19:55.13 | iCEBrkr | xachen: what the heck ya trying to do now?? |
| 19:57.17 | asteriskmonkey | asterisk99: i have lots of experince with the iaxys |
| 19:58.40 | asterisk99 | asteriskmonkey: kewl!!! I have a weird problem with VoiceMailMain().... the users hear <BEEP>-mail instead of "Comedian Mail" |
| 19:59.02 | asterisk99 | asteriskmonkey: This happens no matter how much Wait() I put in |
| 19:59.18 | xachen | I'm trying to stream MMS into my * |
| 19:59.22 | xachen | to play the local radio station |
| 19:59.43 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:59.53 | Druken | use a radio, line in and dsp :) |
| 20:00.25 | Druken | be nice to your damn radio station, save their bandwidth |
| 20:00.40 | asteriskmonkey | asterisk99: your using asterisk@home arnt you! |
| 20:00.53 | xachen | heh |
| 20:00.56 | xachen | problem is |
| 20:00.57 | *** join/#asterisk cool4ever2 (n=craeck@80-218-106-233.dclient.hispeed.ch) |
| 20:00.58 | asterisk99 | asteriskmonkey: No - the full-bore * |
| 20:01.01 | xachen | PBX isn't in radio range :P |
| 20:01.12 | xachen | its only a few thousand miles from the tower |
| 20:01.27 | Druken | then it's not a LOCAL radio station :) |
| 20:01.30 | asteriskmonkey | asterisk99: what version 1.2.1? |
| 20:01.43 | xachen | lol it is :) |
| 20:01.45 | *** join/#asterisk twisty7867 (n=twisty78@adsl-gte-la-216-86-203-111.mminternet.com) |
| 20:01.49 | xachen | its just its cheaper to colo my server out of Canada |
| 20:01.51 | asteriskmonkey | asterisk99: stable or head? |
| 20:02.04 | Druken | xachen: what radio station? |
| 20:02.07 | asterisk99 | asteriskmonkey: CVS-v1-0-10/18/05 |
| 20:02.33 | xachen | Druken: waynefm.com |
| 20:02.39 | xachen | its local customers mainly |
| 20:02.40 | Druken | asteriskmonkey: how sick? like your moms basement kinda sick? |
| 20:02.49 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 20:02.56 | asteriskmonkey | like 151 front street on the backbone sick :D |
| 20:02.58 | Druken | wtf is waynefm... isn't that northbay? |
| 20:03.13 | asteriskmonkey | asterisk99: this problem is recent since you moved from 1.0.9? |
| 20:03.15 | xachen | erm no :) |
| 20:03.23 | xachen | I live in Alberta |
| 20:03.24 | asterisk99 | asteriskmonkey: 151 front is no small potatoes colo |
| 20:03.26 | xachen | if that wil answer all those questins |
| 20:03.36 | xachen | 151 front is the premier colo centre in Canada |
| 20:03.37 | Druken | oh.. alberta ways |
| 20:03.47 | iCEBrkr | xachen: You're wasting your time, CPU and bandwidth trying to do that. |
| 20:03.52 | Druken | 151 is a pain in my ass :) |
| 20:03.54 | asteriskmonkey | me < has 5 cages with 5 redundant connections and torix connection :) |
| 20:03.58 | asterisk99 | asteriskmonkey: dunno - I never used 1.0.9 |
| 20:04.05 | xachen | ToriX |
| 20:04.09 | xachen | evil Nistor :p |
| 20:04.12 | asteriskmonkey | hey now |
| 20:04.20 | asteriskmonkey | he works for rogers you know :P |
| 20:04.24 | xachen | yeah :( |
| 20:04.37 | iCEBrkr | Ding, 3pm.. Nap time |
| 20:04.38 | xachen | I know Myles for the most part form 151 front |
| 20:04.39 | xachen | thats it |
| 20:05.18 | shido6 | who works for rogers? |
| 20:05.35 | asteriskmonkey | nistor :P does nm |
| 20:06.09 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 20:06.20 | asteriskmonkey | asterisk99: upgrade to head :D |
| 20:06.23 | *** join/#asterisk J4k3 (i=j4k3@166.145.106.43) |
| 20:06.45 | Ayano | I just got nvfaxdetect working and it is sending my faxes. How do I change the formatting of the e-mail that it sends? |
| 20:07.39 | *** join/#asterisk rob0 (i=1007@sorry.no-ip-here.net) |
| 20:07.56 | asterisk99 | asteriskmonkey: Is that a known problem with IAXy? |
| 20:08.23 | pegger | why would I not see any udp packets comming out of my asterisk box when I do tcpdump udp??? |
| 20:08.54 | justinu | firewall? |
| 20:09.07 | asteriskmonkey | asterisk99: only heard of it on some dody compiles of aah |
| 20:09.32 | pegger | justinu, but I am running tcpdump on the box that asterisk is on shouldent I see udp packets at least trying to get out |
| 20:09.33 | asterisk99 | asteriskmonkey: dody? aah? |
| 20:09.42 | asteriskmonkey | asterisk99: upgrade to head and see if problem still persists if so let me know ill report it as a bug after some further testing |
| 20:09.49 | asteriskmonkey | aah=asterisk at home |
| 20:09.54 | justinu | pegger: i thoughtt hat tcpdump shows packets on the wire |
| 20:10.31 | justinu | i'm not sure anymore tho |
| 20:10.34 | justinu | forgot |
| 20:10.48 | justinu | anyone care to commend? |
| 20:11.06 | pegger | justinu, yes it shows ethernet packets on eth0 or what ever device you specify, but I woudl think that I would see packets on their way to my router |
| 20:11.17 | pegger | anyone elase have any idea |
| 20:12.12 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 20:12.19 | justinu | pegger: what does iptables -L say? |
| 20:12.34 | pegger | onse sec let me log into the firewall |
| 20:12.46 | asteriskmonkey | asterisk99: if you asterisk -vvvvvvr you should see what happens when you call your voicemail |
| 20:13.21 | asterisk99 | asteriskmonkey: I looked ... no diff (no error msgs) between SIP and IAXy |
| 20:14.01 | justinu | pegger: i mean on your * box |
| 20:14.10 | asteriskmonkey | asterisk99: your not looking for error messages your looking for what happens.. does it go to vm and play a file etc.. |
| 20:14.25 | pegger | justinu, ther is no iptables on the actual box only on the router/firewall |
| 20:15.15 | pegger | justinu, well there are iptables but they are not doing anything |
| 20:15.17 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 20:15.23 | asterisk99 | asteriskmonkey: I see it executing "VoiceMailMain ", but the prompts/ketstrokes are not in the CLI log |
| 20:15.32 | *** join/#asterisk rob0 (i=1007@sorry.no-ip-here.net) |
| 20:15.36 | P4C0 | hello guys, where can I find the complete documentation about register command?? I can't find it in the wiki |
| 20:15.42 | asteriskmonkey | asterisk99: do asterisk -vvvvvvvvvvvvvvvvvvvvvr |
| 20:15.53 | [av]bani | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvr |
| 20:15.57 | asteriskmonkey | you wont see keystokes just numbers dialed and whats happening |
| 20:16.00 | asteriskmonkey | where does it stop |
| 20:16.04 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 20:16.11 | [av]bani | it nevar stops |
| 20:16.28 | pegger | justinu, any ideas? |
| 20:16.47 | P4C0 | r or c? |
| 20:17.06 | Druken | pfft, don't use -vvvvvvvvvvvvvvvvvvvvv, go into the cli, and set verbose ## |
| 20:17.23 | *** join/#asterisk Mrdigital-Work (n=Mrdigita@pool-151-201-148-81.phil.east.verizon.net) |
| 20:17.31 | Mrdigital-Work | can anyone recommend a 1 port pstn card? |
| 20:17.34 | P4C0 | where can I find documentation about register command? |
| 20:17.37 | Mrdigital-Work | for use with asterisk |
| 20:18.15 | Druken | x100p for a single analog pots line |
| 20:18.33 | Mrdigital-Work | x100p clone cards good? |
| 20:18.42 | pegger | P4C0, > http://www.asteriskdocs.org/modules/news/ |
| 20:19.01 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 20:19.14 | asterisk99 | asteriskmonkey: Hmmm... -- Executing VoiceMailMain("IAX2/IAXSOFT1@IAXSOFT1/4", "300@2") in new stack ... 2006-01-03 15:10:30 WARNING[19432]: file.c:550 ast_readaudio_callback: Failed to write frame |
| 20:19.15 | *** join/#asterisk J4k3 (i=j4k3@166.145.106.43) |
| 20:19.24 | Druken | Mrdigital-Work" well, since digium no longer makes the x100p, it's about your only option |
| 20:19.44 | Mrdigital-Work | ok |
| 20:19.51 | Mrdigital-Work | thanks Druken |
| 20:20.03 | P4C0 | pegger, thanks |
| 20:20.04 | memic | i does |
| 20:20.04 | Mrdigital-Work | how much did x100p retail? |
| 20:20.46 | Druken | bout 99usd |
| 20:20.52 | ravenpi | Plug it into Froogle. <$100 |
| 20:20.58 | P4C0 | pegger, nothing |
| 20:21.11 | pegger | P4C0, what? |
| 20:21.17 | asteriskmonkey | asterisk99: looks like mangeled asterisk upgrade and recompile :) |
| 20:21.33 | P4C0 | pegger, register command |
| 20:21.47 | pegger | P4C0, http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:22.08 | pegger | the whole orielly asterisk book in pdf version |
| 20:22.20 | P4C0 | pegger, ok, thanks :D |
| 20:23.47 | [TK]D-Fender | Sangoma voice T3! "Also in this product line is our T3/E3 cards, with full voice support coming in the first quarter of 06." |
| 20:23.56 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167049176.nb.aliant.net) |
| 20:24.40 | eKo1 | t3 cards...yummy |
| 20:26.19 | RoyK | http://www.sofaswitch.com/asterisk.gif |
| 20:26.21 | asteriskmonkey | wish digium would come out with a t3 card |
| 20:26.29 | asteriskmonkey | sangoma .. wanpipe all the way :D |
| 20:26.29 | RoyK | eKo1: wtf would you want a t3 card for? |
| 20:26.52 | asteriskmonkey | for when you run out of pci slots for pri, that and a single t3 is cheaper than a shit load of t1's |
| 20:27.03 | RoyK | if you _purchase_ an asterisk license, it's not meant to support > 4 PRI anyway |
| 20:27.07 | RoyK | 4xT1 |
| 20:27.07 | eKo1 | what else? connect to the pstn and route calls |
| 20:27.10 | RoyK | or E1 |
| 20:27.30 | Druken | ds3 cards for asterisk ? |
| 20:27.34 | Druken | ewhh baby |
| 20:27.39 | RoyK | Druken: I beleive they exist |
| 20:27.44 | mog_work | indeed |
| 20:27.45 | RoyK | Druken: but I wouldn't touch them |
| 20:27.46 | mog_work | nope |
| 20:27.53 | mog_work | no ds3 for asterisk today |
| 20:28.01 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:28.11 | Druken | i dunno if i'd trust a single server for an entire ds3.... |
| 20:28.13 | RoyK | Druken: since more servers with 4 or 8 T1/E1 cards will be better, safer and possibly cheaper |
| 20:28.16 | Druken | but that's me |
| 20:28.22 | RoyK | my point |
| 20:28.33 | Druken | :) |
| 20:28.36 | RoyK | how many T1s are there in a T3? |
| 20:28.41 | pegger | the computer will just be overloaded with 8 t1 lines |
| 20:28.42 | eKo1 | 16 |
| 20:28.42 | RoyK | T3 == DS3, rite? |
| 20:28.47 | RoyK | ok |
| 20:28.53 | mog_work | yes |
| 20:29.08 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 20:29.10 | RoyK | 17 E1s in an E3 |
| 20:29.21 | RoyK | meaning 510 Bchans |
| 20:29.25 | RoyK | without ss7 |
| 20:29.31 | asteriskmonkey | t3 to ds3 is as t1 is to pri |
| 20:29.42 | RoyK | ah |
| 20:29.44 | RoyK | ok |
| 20:29.59 | pegger | what is the diffrence between isdn and idsn pri |
| 20:30.05 | RoyK | isdn is a lot |
| 20:30.09 | eKo1 | you mean bri and pri |
| 20:30.12 | Druken | well, technically a t1 is a ds1 |
| 20:30.23 | RoyK | isdn pri is US isdn over a T1, 42B+D |
| 20:30.24 | Druken | and pri is the primary rate interface :) |
| 20:30.39 | asteriskmonkey | pri is a digital layer that sits ontop a t1 technically :P |
| 20:30.39 | [av]bani | er... there are 30 T1s in a T3 |
| 20:30.42 | eKo1 | isdn pri can go over e1 too |
| 20:30.59 | pegger | so then what is the diffrence between isdn and t1 |
| 20:30.59 | RoyK | [TK]D-Fender: ? |
| 20:31.06 | eKo1 | t1 is a carrier system |
| 20:31.14 | RoyK | pegger: what's the difference between a sofa and an apple? |
| 20:31.14 | eKo1 | isdn is a digital telephony standard |
| 20:31.28 | pegger | so t1 uses isdn |
| 20:31.33 | RoyK | no |
| 20:31.34 | Druken | RoyK: nothing... they can both be green or red |
| 20:31.35 | P4C0 | is there a way to register into a sip provider without the register command?? they guys of my voip provider said that I don't need to register... but I don't know how to deal with that... does anyone know an example about this? |
| 20:31.38 | RoyK | you can run isdn on top of a t1 |
| 20:31.39 | justinu | 28 DS1s on a DS3 |
| 20:31.40 | [av]bani | telco people usually say T3 when they mean channelized, and DS3 when they mean unchannelized (eg atm) |
| 20:31.40 | RoyK | or an e1 |
| 20:31.43 | justinu | not 30 |
| 20:31.48 | [TK]D-Fender | RoyK : ? |
| 20:32.15 | justinu | T3 is actually referring to a physical standard, cables, connectors, etc. |
| 20:32.19 | justinu | DS3 is just the data rate |
| 20:32.25 | [av]bani | and framing... |
| 20:32.28 | justinu | yes |
| 20:32.56 | RoyK | and colour |
| 20:33.03 | [av]bani | Interface ATM1/0 is up |
| 20:33.03 | [av]bani | Hardware is ENHANCED ATM PA - DS3 (45000Kbps) |
| 20:33.03 | [av]bani | Framer is PMC PM7345 S/UNI-PDH, SAR is LSI ATMIZER II |
| 20:33.03 | [av]bani | Firmware rev: G153, Framer rev: 1, ATMIZER II rev: 3 |
| 20:33.07 | [av]bani | :) |
| 20:33.16 | RoyK | what colour? |
| 20:33.22 | [av]bani | yellow |
| 20:33.26 | RoyK | hm |
| 20:33.30 | RoyK | should be red |
| 20:34.27 | *** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net) |
| 20:36.29 | *** join/#asterisk viperdude (n=viperdud@84-45-168-57.no-dns-yet.enta.net) |
| 20:37.22 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 20:39.51 | paryl | i've had nothing but issues with agents and queues... i can't get autologoff or timeout to work correctly |
| 20:40.13 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 20:40.40 | rculp | is anyone familiar with the directory function? |
| 20:40.54 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 20:41.01 | shido6 | yes |
| 20:41.18 | shido6 | looking for voicemail.conf? |
| 20:41.45 | rculp | well, I got it successfully doing that |
| 20:41.50 | rculp | but get an error |
| 20:41.53 | rculp | when it attempts to dial |
| 20:42.00 | rculp | I know I missed something simple |
| 20:42.05 | rculp | let me pastebin the error |
| 20:42.07 | shido6 | to dial what... |
| 20:42.27 | rculp | http://pastebin.ca/35648 |
| 20:42.44 | rculp | the extension they find via directory |
| 20:42.48 | *** join/#asterisk nahuel_ (n=nahuel@OL33-83.fibertel.com.ar) |
| 20:43.03 | rculp | exten => 5,1,Directory(default,incoming,f) |
| 20:43.04 | shido6 | muahahah |
| 20:43.09 | rculp | is what I have in my extensions.conf |
| 20:43.24 | rculp | and I have all extensions under the internal context |
| 20:43.32 | rculp | which is included under incoming |
| 20:43.55 | rculp | so I think there is just one thing I'm missing |
| 20:44.01 | rculp | looking in the wiki |
| 20:44.48 | *** join/#asterisk javar (n=javar@69.79.133.185) |
| 20:44.54 | javar | hi |
| 20:45.12 | javar | Can anybody help with the algorithm to extract the country and city from a call detail record? |
| 20:45.22 | *** join/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net) |
| 20:45.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:45.30 | *** part/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net) |
| 20:46.04 | Mrdigital-Work | http://cgi.ebay.com/UNLOCKED-NEW-LINKSYS-PAP2-VOIP-2-PORTS-Sipura-SPA-2k_W0QQitemZ5848472040QQcategoryZ61840QQssPageNameZWDVWQQrdZ1QQcmdZViewItem can this be used with analog phones and asterisk? |
| 20:48.31 | [TK]D-Fender | Mrdigital-Work : If its unlocked, yes. Pray they aren't mistaken about that and expect the price to go up. |
| 20:49.06 | javar | Can anybody help with the algorithm to extract the country and city from a call detail record? |
| 20:49.40 | [TK]D-Fender | Mrdigital-Work, I'd suggest you just go and buy an SPA-2002 and be sure of it. |
| 20:50.22 | [TK]D-Fender | Mrdigital-Work : Also note their terms "* Standard Flat Rate Shipping Service: US $14.99 " |
| 20:51.19 | *** join/#asterisk Tili (i=Tili@202-133-67-166-dialup.sat.net.pk) |
| 20:52.10 | *** join/#asterisk JSingle (n=Johnny@S01060004e2c23df8.wp.shawcable.net) |
| 20:52.39 | asteriskmonkey | ive got the linksys pap2 unlock hack |
| 20:52.49 | *** join/#asterisk darby_t (i=darby_t@dla169.neoplus.adsl.tpnet.pl) |
| 20:52.49 | Mrdigital-Work | hack? |
| 20:52.55 | justinu | give me an RTP300 unlock, and i'll be impressed |
| 20:53.06 | asteriskmonkey | yes rewrites the pap2 bios so you can use it on any provider and asterisk |
| 20:53.27 | asteriskmonkey | justinu: give me an rtp300 and a couple weeks :) |
| 20:53.36 | Mrdigital-Work | can you plug a splitter in the fxs adapter? |
| 20:53.38 | *** join/#asterisk implicit (n=implicit@65.165.85.44) |
| 20:53.53 | asteriskmonkey | Mrdigital-Work sure but you still only get 1 channel |
| 20:53.59 | Mrdigital-Work | 1 chanel? |
| 20:54.16 | Mrdigital-Work | ohhhh |
| 20:54.20 | Mrdigital-Work | you cant call other phones |
| 20:54.20 | asteriskmonkey | well if its an fxs port and you put a splitter on it its like splitting a single line phone |
| 20:54.21 | Mrdigital-Work | gotcha |
| 20:54.30 | JSingle | hey, is there a way to use asterisk and an old modem with a phone to make voip calls? |
| 20:54.35 | Mrdigital-Work | jsingle |
| 20:54.38 | Mrdigital-Work | there was |
| 20:54.39 | asteriskmonkey | yes |
| 20:54.47 | Mrdigital-Work | they removed it in 1.2 |
| 20:54.52 | asteriskmonkey | infact certain old modems are used to make the x100p |
| 20:55.11 | asteriskmonkey | really gone in 1.2? |
| 20:55.17 | Mrdigital-Work | chanspy is out |
| 20:55.24 | JSingle | is there a tut that tells you how to do this |
| 20:55.27 | Mrdigital-Work | asteriskmonkey? make? |
| 20:55.27 | asteriskmonkey | mmm probably just have to fuss with if .c files and recompile em |
| 20:56.03 | asteriskmonkey | Mrdigital-Work: yes a few older modems had 2 specific chipsets which you could flash to show up as x100ps |
| 20:56.13 | asteriskmonkey | there is a link on the voip wiki somewhere about that |
| 20:56.16 | Mrdigital-Work | do you know the chipset? |
| 20:56.24 | Mrdigital-Work | can i just get a FXS Card? |
| 20:56.27 | JSingle | no i dont |
| 20:56.30 | asteriskmonkey | cant remeber of hand searcht the voip wikick |
| 20:56.38 | JSingle | ok |
| 20:56.43 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 20:56.46 | asteriskmonkey | Mrdigital-Work just use an iaxy thats a simple fxs |
| 20:57.04 | asteriskmonkey | <PROTECTED> |
| 20:57.14 | asteriskmonkey | <PROTECTED> |
| 20:57.17 | *** join/#asterisk PakiPenguin_ (i=uppal@linuxpakistan/admin/pakipenguin) |
| 20:57.18 | g__ | it's the ISP, not us. |
| 20:57.20 | Mrdigital-Work | iaxy? |
| 20:57.37 | asteriskmonkey | <PROTECTED> |
| 20:57.41 | asteriskmonkey | lol |
| 20:57.44 | asteriskmonkey | ah |
| 20:57.45 | asteriskmonkey | ok |
| 20:57.56 | g__ | That's a good one.. I should go make kids or something. |
| 20:58.12 | asteriskmonkey | g__ check if its transmitting in full duplex or half |
| 20:58.43 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 20:58.43 | Mrdigital-Work | <PROTECTED> |
| 20:58.46 | g__ | Again, it's the ISP: I think the line's being shared with someone else in the building. |
| 20:58.50 | Mrdigital-Work | thats the clones |
| 20:59.13 | g__ | Actually, I *know* someone else in the building also has fibre.. |
| 20:59.23 | asteriskmonkey | well take it up with the isp :D |
| 20:59.53 | [TK]D-Fender | Mrdigital-Work : If you wan FXS that'd be either a TDM400P or an external ATA. |
| 21:00.08 | g__ | Good idea. Am I still allowed to grumble on the channel? |
| 21:00.27 | ast_freak | g__: no. |
| 21:00.34 | g__ | Oh... |
| 21:00.46 | asteriskmonkey | the TDM400P you can load up with up to 4 daugther boards wither fxo/fxs or any combo there of |
| 21:01.29 | g__ | ast_freak can be cruel when he's sober.. |
| 21:06.34 | *** part/#asterisk Splas (i=jwb@206.252.198.100) |
| 21:08.36 | *** join/#asterisk SugarGuest604 (n=SugarGue@mail.singlepointnetworks.com) |
| 21:09.24 | [TK]D-Fender | Though for the money (and functionality) Id much sooner suggest Sipura ATA's |
| 21:09.36 | *** join/#asterisk Martz (n=martz@81.6.250.233) |
| 21:10.22 | xheliox | Anyone have any advice on this issue... making a call, IAX2 to Zap, there seems to be an annoying hiss whenever the call connects to Zap. It's fine if I call Zap to Zap on the same box. |
| 21:10.24 | xheliox | Any thoughts? |
| 21:11.43 | Mrdigital-Work | asteriskmonkey: can ihave the pap2 hack? |
| 21:12.50 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 21:13.51 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 21:15.57 | Mrdigital-Work | can i setup the phones to connect to a asterisk box over the internet/ |
| 21:16.19 | *** join/#asterisk rene- (n=rene-@201.144.61.144) |
| 21:16.58 | *** join/#asterisk mcquaid (i=mcquaid@toronto-hs-216-138-233-79.s-ip.magma.ca) |
| 21:17.16 | eKo1 | yes |
| 21:17.30 | mcquaid | hello, i haven't used asterisk yet, well i havent' even installed it yet but here's what i'm looking for |
| 21:18.18 | mcquaid | i have a voip provider which service quality is acceptable but they have a voicemail option that encodes the voicemail and then emails it to an acct |
| 21:18.35 | Mrdigital-Work | ok |
| 21:18.45 | rene- | Hey all, i am doing some research about large profile asterisk installations, i ve gone tru both the wiki and the digium.com web pages but the names of the companies involved in the case studies are not very well known, does anyone know about large asterisk installations for high profile costumers and would like to talk about it? |
| 21:19.05 | mcquaid | however for whatever reason, there voicemail quailty is horrible so waht i want to do is record the voicemail locally and intercept the call before it goes to their vociemail server |
| 21:19.06 | [hC] | doh.. so, how do i solve this double ring problem? :P when i dial out my pri, i get ringing from asterisk, as well as ringing from the pri i think. I remember reading about this, but i forget the solution |
| 21:19.11 | *** join/#asterisk backblue (n=moo@87-196-4-173.net.novis.pt) |
| 21:19.14 | [hC] | something about telling asterisk not to send ringing? |
| 21:19.39 | rene- | you just have to take out the 'r' in your dial statement |
| 21:19.41 | mcquaid | so this is just for using with one sip acct, not managing a phone network or anything just for personal use on the one box |
| 21:20.04 | mcquaid | would asterisk be appropriate for this? and what are the requirements of the voip client to interface with asterisk? |
| 21:20.06 | *** part/#asterisk Chonlada (i=somjuk@jane.lru.ac.th) |
| 21:20.39 | iCEBrkr | mcquaid: asterisk would be overkill, but people like me do it anyhow :) |
| 21:21.05 | mcquaid | ya i thought it would be overkill, but haven't found any other option for voicemail recording with voip |
| 21:21.15 | mcquaid | no client i know of anyways has built in voice recording |
| 21:21.40 | mcquaid | can any voip/sip client interface with asterisk such as twinkle linphone? |
| 21:21.41 | cypromis | <PROTECTED> |
| 21:22.01 | mcquaid | and would asterisk run fine on the same box as the sip client? |
| 21:22.20 | [TK]D-Fender | mcquaid : X-Pro / eyeBeam does VM in its softphone. |
| 21:22.21 | iCEBrkr | mcquaid: sure |
| 21:23.08 | mcquaid | [TK]D-Fender, thx for the suggestions, i'll look into those clients, hopefully they'll run in wine |
| 21:23.26 | iCEBrkr | There's always some one who can't just do things the normal way. |
| 21:23.32 | mcquaid | iCEBrkr, is this a daunting task to set up asterisk for one local client and just to record voicemail? |
| 21:23.54 | mcquaid | [TK]D-Fender, are both those free? |
| 21:24.01 | iCEBrkr | mcquaid: for what you described, it's entirely too much work for what you wanna do :) |
| 21:24.35 | [TK]D-Fender | mcquaid : NEITHER |
| 21:24.36 | mcquaid | heh, ya i thought it might but i might not have any other option |
| 21:24.36 | iCEBrkr | mcquaid: If you're hardcore about wanting to learn VoIP and do more than just voicemail... sure it'd be worth the hassle. |
| 21:24.45 | mcquaid | and besides, might be kinda fun |
| 21:25.09 | mcquaid | also, i assume once it's set up i do neat things like customized greetings based on called id etc |
| 21:26.39 | mcquaid | ok, so just to make sure this is possible, can any sip client work with asterisk? or are there special requirements of the sip client software? |
| 21:27.23 | mcquaid | my favourite so far linux wise is twinklephone |
| 21:28.08 | *** join/#asterisk hardwire (n=nnnhardw@66-230-102-166-cdsl-rb1.nwc.acsalaska.net) |
| 21:28.10 | hardwire | hmm |
| 21:28.15 | hardwire | I should mirror voip-info |
| 21:28.36 | iCEBrkr | mcquaid: If it speaks SIP then it's highly likely to work with Asterisk. |
| 21:28.47 | freestyle_networ | any conferencing gurus in the house? |
| 21:31.05 | mcquaid | iCEBrkr, thx, how resource hungry is asterisk in managing just one acct on the same box? |
| 21:31.28 | iCEBrkr | it's not |
| 21:31.37 | mcquaid | cool |
| 21:31.53 | mcquaid | i guess i should get ready to start pulling my hair out and try and set this up ;) |
| 21:31.58 | iCEBrkr | I run it on my 1.3Ghz machine that is also my web, mail, teamspeak and mysql server |
| 21:32.07 | iCEBrkr | mcquaid: Just install it.. Don't be scared to try things. |
| 21:32.30 | mcquaid | ok, but again this box is my desktop as well, thats why i asked |
| 21:32.41 | iCEBrkr | So it's your desktop... |
| 21:32.44 | iCEBrkr | ?? |
| 21:33.03 | iCEBrkr | You think this is some sort of bloated windows application? |
| 21:33.42 | iCEBrkr | I've been testing on a 700mhz AMD machine with 512megs of ram for the past 3 months... It works just fine-- even doing VoIP |
| 21:35.04 | *** join/#asterisk CPC-BR (n=bdcfl@201.29.156.32) |
| 21:35.19 | eKo1 | i have asterisk 1.0 and 1.2 running |
| 21:35.27 | eKo1 | postgres 7.4 and 8.1 running |
| 21:35.35 | eKo1 | apache 1.3 and 2 running |
| 21:35.40 | eKo1 | all on my desktop machine |
| 21:35.59 | iCEBrkr | Hey, how about you make it even MORE complex and install a few versions of MySQL |
| 21:36.02 | eKo1 | it has 512 mb of ram |
| 21:36.04 | Ayano | does anyone know where I can change the from e-mail addresses on outgoing faxes? |
| 21:36.12 | justinu | lol |
| 21:36.16 | eKo1 | actually, i have mysql also |
| 21:36.25 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:36.47 | lunk | eKo1: if you aren't in KDE, you just aren't cool. |
| 21:36.52 | iCEBrkr | eKo1: All that really sounds like the intelligent way to do things... |
| 21:37.02 | iCEBrkr | </sarcasm> |
| 21:37.04 | eKo1 | kde? |
| 21:37.07 | eKo1 | no no no |
| 21:37.16 | lunk | well you have to use that swap space! |
| 21:37.23 | lunk | all of it ;) |
| 21:37.44 | CPC-BR | helo everyone, I have a problem to connect 2 * servers using SIP. The follow error is presented (== Everyone is busy/congested at this time) |
| 21:37.55 | CPC-BR | who know how to solve this problem?? |
| 21:38.00 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 21:38.03 | eKo1 | well, someone is busy |
| 21:38.08 | eKo1 | big deal |
| 21:38.35 | eKo1 | i get that all that time when someone tries to place a call to a busy line |
| 21:39.02 | *** join/#asterisk implicit (n=implicit@65.165.85.44) |
| 21:39.05 | eKo1 | oh, you could also be getting it because the configuration is bad |
| 21:39.24 | iCEBrkr | eKo1: That's pretty helpful |
| 21:39.25 | eKo1 | like say the host= part in the sip.conf |
| 21:39.35 | eKo1 | is wack |
| 21:39.41 | *** part/#asterisk rene- (n=rene-@201.144.61.144) |
| 21:39.43 | iCEBrkr | Thanks for stating the obvious :) |
| 21:39.59 | eKo1 | yep, most problems have obvious solutions |
| 21:40.04 | CPC-BR | ok but what could be bad?? |
| 21:40.27 | iCEBrkr | eKo1: I wouldn't say that... I'd say something like "Most problems have solutions that can be found on the Wiki" |
| 21:41.47 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 21:42.14 | Hmmhesays | anyone subscribed to the dev list got the playdialtone patch? |
| 21:42.59 | CPC-BR | wiki?? |
| 21:43.07 | CPC-BR | do u have the url? |
| 21:43.14 | iCEBrkr | ~docs |
| 21:43.17 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 21:43.56 | *** join/#asterisk _-_ (n=nabudoco@ns1.ensenada.gob.mx) |
| 21:44.49 | *** part/#asterisk SugarGuest604 (n=SugarGue@mail.singlepointnetworks.com) |
| 21:45.26 | CPC-BR | hey guys when have I to use "insecure"?? |
| 21:45.36 | *** join/#asterisk prh (n=paul@212.13.203.80) |
| 21:45.47 | justinu | when the peer doesn't feel like doing a proxy authentication |
| 21:46.18 | CPC-BR | could u explain better?? |
| 21:46.23 | justinu | not really |
| 21:46.41 | CPC-BR | or in a dif. way :) |
| 21:46.42 | justinu | it means that the other side doesn't feel like authenticating with * |
| 21:46.58 | justinu | so * just accepts its calls |
| 21:47.55 | CPC-BR | could u see the entire error in pvt?? |
| 21:48.21 | CPC-BR | I'm becomin crazy with this error n I need to solve it quickly :) |
| 21:48.24 | CPC-BR | lol |
| 21:49.17 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
| 21:51.47 | CPC-BR | what does the folowing error means?? create_addr: No such |
| 21:51.47 | CPC-BR | host: |
| 21:52.26 | CPC-BR | what does the folowing error means?? |
| 21:52.26 | CPC-BR | "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack |
| 21:52.27 | CPC-BR | Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such |
| 21:52.27 | CPC-BR | host: 10.0.0.121/100 |
| 21:52.27 | CPC-BR | Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to |
| 21:52.29 | CPC-BR | create channel of type 'SIP' |
| 21:52.31 | CPC-BR | == Everyone is busy/congested at this time |
| 21:54.34 | eKo1 | i knew it, it is a host= problem |
| 21:54.53 | *** join/#asterisk CPC-BR (n=bdcfl@201.29.156.32) |
| 22:00.00 | Mrdigital-Work | what was eKo1 |
| 22:02.30 | [hC] | Interesting. |
| 22:02.51 | [hC] | I'm getting this double-ringing issue when dialing out my pri, yet i dont have the 'r' option passed tr dial. |
| 22:03.01 | [hC] | seems to only happen on particular types of sip phones, too... |
| 22:04.08 | [hC] | oh wait. maybe i do. |
| 22:04.12 | *** join/#asterisk Utah_Dave (n=boucha@0-2pool130-207.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:04.45 | [hC] | hmm. nope. only when dialing internal extensions. |
| 22:04.45 | [hC] | wtf mate. |
| 22:04.46 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:05.25 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 22:05.37 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 22:10.02 | CPC-BR | is easy to link 2 * servers using SIP? and how to do that?? |
| 22:10.27 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 22:10.42 | *** join/#asterisk freestyle_networ (n=chatzill@68.148.192.184) |
| 22:11.14 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 22:11.18 | *** join/#asterisk rob0 (i=1007@sorry.no-ip-here.net) |
| 22:12.12 | P4C0 | hello guys, is there a way to force asterisk to use an ip address as his address?? I'm inside a private lan, but my gateway is forwarding the ports to my private ip address, how can I give the public address to asterisk so he take it as his address? externip= dosen't seem to work |
| 22:12.41 | Nugget | externip is how you do that. |
| 22:12.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:13.31 | trixter | did you *also* set localnet? |
| 22:13.35 | CoaxD | P4C0: You really shouldnt do that |
| 22:13.35 | trixter | you need both |
| 22:13.44 | P4C0 | Nugget, in the general context? (in sip.conf?) do I need to set localnet? or nat? |
| 22:13.49 | P4C0 | CoaxD, why? |
| 22:14.00 | CoaxD | P4C0: If the address is NAT, give it the nat address. not the public |
| 22:14.08 | CoaxD | P4C0: it can cause arp problems. |
| 22:14.42 | CoaxD | P4C0: (The external interface knows about the external addresses. the internal interface knows about internal addresses. You cant just ifconfig the public address internally.) |
| 22:15.13 | P4C0 | CoaxD, so, how should I do it? |
| 22:15.42 | CoaxD | P4C0: You should A) put * outside the firewall, or B) ifconfig the private static address and blindly forward everything on the public to everything on the internal |
| 22:16.04 | CoaxD | P4C0: but that does kinda break the need for asterisk on the internal net anyway |
| 22:16.07 | P4C0 | CoaxD, A is not an option (I only have 1 public ip address) |
| 22:16.32 | CoaxD | P4C0: Ahhhhh. i see. Well, option B is the way you'd do it |
| 22:16.36 | P4C0 | CoaxD, I didn't get the option B... |
| 22:16.40 | CoaxD | P4C0: Uh |
| 22:16.52 | CoaxD | P4C0: You just cant ifconfig the external IP when plugged into the internal network, man |
| 22:17.05 | *** join/#asterisk pablasso (n=pablasso@dsl-200-78-96-203.prod-infinitum.com.mx) |
| 22:17.07 | CoaxD | P4C0: You need to ifconfig the internal IP and portforward everything to it from the public - via your firewall |
| 22:17.14 | pablasso | hi people |
| 22:17.26 | P4C0 | CoaxD, what do you mean by ifconfig the internal ip ? |
| 22:17.42 | pablasso | any of you have used chanspy? or something else recommended to be able to record calls? |
| 22:17.48 | Hmmhesays | nothing like manually patching stuff in |
| 22:17.49 | CoaxD | P4C0: You need to go back to tcp/ip school, man :) |
| 22:17.51 | P4C0 | CoaxD, with my firewall I have port 5060 and 10000 to 10500 forwared to my private ip address |
| 22:18.03 | CoaxD | p4c0: Okay. well, thats not what you asked |
| 22:18.10 | P4C0 | CoaxD, no, I just don't understand what you are trying to say |
| 22:18.28 | CoaxD | p4c0: Ahhh, I see exactly what you meant |
| 22:18.32 | CoaxD | p4c0: nat=yes |
| 22:18.35 | CoaxD | p4c0: That'll do it |
| 22:18.40 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:18.46 | *** join/#asterisk Martz` (n=martz@62.3.201.10) |
| 22:19.07 | P4C0 | CoaxD, ok, but then I have to put nat=yes and also externip and localnet? or only nat=yes?that in the general context right? |
| 22:19.50 | pablasso | or anyone knows if ChanSpy works on asterisk 1.2? |
| 22:20.23 | P4C0 | CoaxD, ? |
| 22:21.00 | CoaxD | p4c0: I've never actually had to do anything other than nat=yes |
| 22:21.08 | P4C0 | CoaxD, thanks |
| 22:21.17 | *** join/#asterisk CPC-BR (n=bdcfl@201.29.156.32) |
| 22:21.33 | CoaxD | nat=yes |
| 22:21.34 | CoaxD | canreinvite=no |
| 22:21.34 | CoaxD | qualify=200 |
| 22:21.37 | CoaxD | make sure those things are set |
| 22:21.40 | CoaxD | and you should be all good |
| 22:21.58 | CoaxD | (thats in sip.conf) |
| 22:22.19 | *** join/#asterisk swm__ (n=root@digitaldatabits.net) |
| 22:22.34 | P4C0 | CoaxD, thanks |
| 22:22.51 | ManxPower | externip and localnet is only if ASTERISK is behind NAT |
| 22:23.03 | [hC] | you may need externip and localnet as well if asterisk is advertising itself as the internal IP |
| 22:23.25 | swm__ | Wow a nice irc client ... |
| 22:23.29 | [hC] | therefore the remote clients are trying to answer to your internal address |
| 22:23.42 | [hC] | externip rewrites the ip in the SIP header so that they reply to the correct address |
| 22:24.09 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:24.14 | ManxPower | pablasso, "show applications like monnitor" in the asterisk CLI |
| 22:24.26 | [hC] | So, anyone seen a double-ring issue going SIP->IAX->PRI when you DONT specify 'r' as an argument to Dial() ? |
| 22:24.29 | P4C0 | :) |
| 22:24.38 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 22:25.31 | P4C0 | what does the qualify means? |
| 22:25.46 | *** part/#asterisk Da-TimE-BoMB (n=root@digitaldatabits.net) |
| 22:26.17 | [hC] | qualify=yes means that it will check the latency of the peer, and if it exceeds a certain amount, mark it as unreachable |
| 22:26.22 | [hC] | I think the default is 2000 or something |
| 22:26.35 | [hC] | if you set qualify=500 for example, if it exceeded 500ms it would mark it as unreachable |
| 22:26.48 | [hC] | also, using sip show peers, it will show you the current ms latency, as opposed to saying 'Unmonitored' |
| 22:27.17 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
| 22:28.03 | P4C0 | [hC], thanks |
| 22:30.14 | [hC] | no problem. |
| 22:33.12 | pablasso | ManxPower, i use monitor already, but since i want to record a call wich has already started, i think i need something like ChanSpy to be able to record it? or theres another way i could do that? |
| 22:33.39 | distortion | [hc] does qualify=yes use icmp? |
| 22:33.52 | xheliox | distortion: No. |
| 22:34.09 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 22:35.01 | distortion | so, ideally it should qualify the sip ip/port that a peer is registered on right? |
| 22:35.05 | ManxPower | pablasso, no. see features.conf.sample in the src/asterisk/configs/ directory |
| 22:35.37 | [hC] | distortion: it sends its qualifies via the sip ip/port the peer is connected on yes. |
| 22:35.52 | ManxPower | distortion, qualify=yes makes Asterisk send a SIP OPTIONS packet to the SIP device, the SIP device responds and asterisk measures the time it takes. |
| 22:35.52 | Cyberchen | is there someone out there who uses bristuff+AMP ? |
| 22:36.14 | *** join/#asterisk seele_ (n=seele@200.124.172.72) |
| 22:36.30 | pablasso | MaxxPower, ill take a look at it, thank you |
| 22:36.39 | seele_ | please help with gnugk |
| 22:36.42 | distortion | thx guys. |
| 22:36.46 | P4C0 | ok, that was weird... I need help... |
| 22:37.14 | seele_ | some channel for gnugk or oh323 |
| 22:37.36 | P4C0 | I just called my sip provider asking about one problem that i have with the re-register, and they just said ok, comment the registry line... I can accept calls and make them without registering... how is that possible!? |
| 22:37.40 | ManxPower | seele_, The H323 channel driver included in asterisk-addons is not good for you. |
| 22:37.59 | ManxPower | P4C0, it isn't once your IP address changes. |
| 22:38.08 | [av]bani | chan_h323 is no good, yep |
| 22:38.13 | [av]bani | need oh323 :/ |
| 22:38.41 | P4C0 | ManxPower, sorry? |
| 22:38.43 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 22:38.52 | seele_ | [av]bani, yes oh323 channel please |
| 22:39.24 | *** join/#asterisk ToTo (n=ToTo@host125-131.pool872.interbusiness.it) |
| 22:39.50 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
| 22:40.35 | asterisk99 | anyone know how to turn ADSI signals to IAXy phones off?? |
| 22:40.51 | seele_ | ManxPower, yes I install it but i need some hep with the configuration |
| 22:42.02 | asterboy | What is the policy for jbot additions/corrections? Is it being administrated by ops only? |
| 22:42.09 | justinu | anyone can do it |
| 22:42.21 | asterboy | what is the syntax? |
| 22:42.30 | asterboy | ~jbot |
| 22:42.31 | jbot | jbot is, like, only marginally useful at best, or a silly little bugger |
| 22:42.38 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 22:42.55 | justinu | jbot, asterboy is a guy who wants to figure out how to use you |
| 22:42.57 | jbot | justinu: okay |
| 22:43.01 | justinu | ~asterboy |
| 22:43.02 | jbot | [asterboy] a guy who wants to figure out how to use you |
| 22:43.07 | *** join/#asterisk Muiz (i=someone@dhcp185-1-186.dsl.ucc-net.ca) |
| 22:43.10 | Hmmhesays | heyo |
| 22:43.15 | Hmmhesays | anyone know what this changed to? pbx.c: In function `__ast_pbx_run': |
| 22:43.15 | Hmmhesays | pbx.c:2366: error: structure has no member named `writeinterrupt' |
| 22:43.17 | *** part/#asterisk Muiz (i=someone@dhcp185-1-186.dsl.ucc-net.ca) |
| 22:43.17 | asterboy | thanks! :P |
| 22:43.29 | justinu | job, no, asterboy is a guy who now knows how to use you |
| 22:43.42 | justinu | jbot, has there been any thing new on adding payloads to control frames in the past month or so? |
| 22:43.44 | justinu | oops |
| 22:43.50 | justinu | jbot, no, asterboy is a guy who now knows how to use you |
| 22:43.52 | jbot | justinu: okay |
| 22:43.56 | justinu | ~asterboy |
| 22:43.57 | jbot | [asterboy] a guy who now knows how to use you |
| 22:44.06 | justinu | hah, you get the idea |
| 22:44.19 | *** join/#asterisk saitech (n=admin@85.235.237.14) |
| 22:44.27 | Seldon1975 | wtf? |
| 22:45.01 | justinu | Hmmhesays: sounds like a possible bug |
| 22:45.03 | saitech | can anyone tell me, how to do call limiting for an agent/sippeer in a queue? It doesnt function with call-limit(incominglimit) in asterisk 1.2.1 but it did in asterisk 1.0.7 |
| 22:45.13 | justinu | Hmmhesays: what circumstances are you getting that in? |
| 22:45.35 | *** part/#asterisk Utah_Dave (n=boucha@0-2pool130-207.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:46.31 | Hmmhesays | i'm assuming waitinterrupt changed to someting else, cause i patched 1.21 with an old patch that makes dialtone stop playing when the first digit is recieved |
| 22:46.56 | justinu | Hmmhesays: oh, in that case... not sure what to tell you |
| 22:47.00 | *** join/#asterisk ejr (n=ed@c-67-185-13-136.hsd1.wa.comcast.net) |
| 22:47.08 | justinu | maybe one of the devlopers can help you |
| 22:47.19 | justinu | or post it on mantis ask for a new patch? |
| 22:47.28 | Hmmhesays | +if (c->writeinterrupt) |
| 22:47.29 | Hmmhesays | +ast_deactivate_generator(c); |
| 22:47.52 | Hmmhesays | those are the lines that get patched into pbx.c ast_deactive_generator seems to be a valid function |
| 22:48.05 | justinu | yeah, obviously writeinterrupt is gone |
| 22:48.19 | Hmmhesays | but what did it get replaced with |
| 22:48.26 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 22:49.55 | saitech | can anyone tell me, how to do call limiting for an agent/sippeer in a queue? It doesnt function with call-limit(incominglimit) in asterisk 1.2.1 but it did in asterisk 1.0.7. Is it possible with GROUPCOUNT() ? |
| 22:52.44 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 22:53.11 | Dr-Linux | from where can i edit/change the voicemail messages ? |
| 22:55.50 | *** join/#asterisk OloBola (n=not@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
| 22:55.51 | distortion | saitech: try: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup |
| 22:56.01 | Dr-Linux | from where can i edit/change the voicemail messages ? |
| 22:56.08 | distortion | there are several options there, let me know if you get it to work |
| 22:56.16 | *** join/#asterisk implicit (n=implicit@200.12.227.205) |
| 22:56.26 | pablasso | Manxpower, i already saw the features.conf.sample but didnt saw anything useful on it to be able to record a call that is already bridged... or i misunderstood? |
| 22:57.38 | wunderkin | manually record it, you are the one on the call? |
| 22:57.46 | distortion | Dr: be more specific, voicemail greeting messages or actual voicemails? |
| 22:58.36 | *** join/#asterisk roulduke_ (i=z9ebpmcz@p508D10C4.dip0.t-ipconnect.de) |
| 22:58.43 | ejr | Hello all. Quick question. To get 1.2.1 from SVN. Do I use the 'svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2' command? |
| 22:59.17 | malverian[work] | I have a reproducible crash with Asterisk-1.2.1 here |
| 23:03.04 | *** join/#asterisk _cleric_ (n=dacleric@p5482ACEC.dip0.t-ipconnect.de) |
| 23:03.19 | justinu | malverian[work]: what's going on? |
| 23:04.01 | malverian[work] | We have a buggy soft phone that causes the server to crash.. when trying to delete a non-existant scheduled event. |
| 23:04.16 | justinu | ah |
| 23:04.28 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 23:04.56 | ManxPower | pablasso, it's not in 1.0, only in 1.2 |
| 23:05.12 | ManxPower | specifically the one touch record. It's also been discussed on the mailing list in the past 2 months |
| 23:06.13 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 23:06.36 | pablasso | ManxPower, thanks im gonna try with it |
| 23:07.57 | [hC] | malverian[work]: enter it on bugs.digium.com, it will get attention within an hour or two |
| 23:08.14 | *** join/#asterisk fulco (n=fulco1@d-ip-129-15-10-85.ucs.ou.edu) |
| 23:08.51 | *** join/#asterisk JSingle (n=Johnny@S01060004e2c23df8.wp.shawcable.net) |
| 23:09.30 | JSingle | Foreign Exchange Station And Foreign Exchange Office???? |
| 23:09.52 | justinu | i always thought it was foreign exchange subscriber, but ok :) |
| 23:10.49 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 23:11.06 | Dr-Linux | distortion: actual message, like when press 9999 it says "password" i wanna change this password voice file? from where can i change it? |
| 23:11.17 | JSingle | Ok, But If I Have A Modem In My Computer What One Is It Or CAn A Modem Do Both ? |
| 23:12.33 | Beirdo | Let's Capitalize Every Word... |
| 23:12.38 | Beirdo | is this a song title? |
| 23:12.41 | JSingle | sry |
| 23:12.42 | justinu | Dr-Linux: ls -l /var/lib/asterisk/sounds/vm* |
| 23:12.46 | *** join/#asterisk ejr (n=ed@c-67-185-13-136.hsd1.wa.comcast.net) |
| 23:12.55 | Beirdo | your modem is a modem |
| 23:13.13 | asterisk99 | Are there any Asterisk source code gurus here? :) |
| 23:13.17 | Beirdo | in very specific cases, some modems can be used as FXO interfaces |
| 23:13.29 | Beirdo | but not many of them, and it doesn't always work well |
| 23:13.36 | JSingle | yes but if i want to connect a phone for voip i heard i can do it with a modem? |
| 23:13.55 | Dr-Linux | ooo okey justinu thanks let me check |
| 23:13.57 | justinu | xp100? |
| 23:14.00 | justinu | x100p? |
| 23:14.02 | justinu | something like that |
| 23:15.10 | Dr-Linux | :S |
| 23:16.00 | Dr-Linux | justinu: i wanna change this message: when it says "password" after pressing 9999 |
| 23:16.00 | Dr-Linux | i want it like "please enter your password" |
| 23:16.02 | Dr-Linux | so i can't find this "password" vm |
| 23:16.02 | justinu | you see the file called vm-password.gsm? |
| 23:16.09 | Dr-Linux | no |
| 23:16.09 | justinu | i bet you 20 bucks that's the prompt |
| 23:16.39 | justinu | ls -l /var/lib/asterisk/sounds/vm-password.gsm |
| 23:17.00 | Dr-Linux | its not there |
| 23:17.10 | justinu | that's where it is on my box |
| 23:17.34 | distortion | dr: run "updatedb" then "locate vm-password" |
| 23:17.56 | freestyle_networ | anyone know of any benefits to using a hardware timmer (xp100 lets say) versus ztdummy ? |
| 23:18.54 | Dr-Linux | justinu: thanks dude i just got it .. but all i need now .. to find "please enter your password" message :) |
| 23:19.22 | justinu | Dr-Linux: yeah, you'll have to record it yourself, or find it somehow |
| 23:19.28 | iDunno | or look in /usr/share/asterisk/sounds |
| 23:19.41 | iDunno | and you can pay for it from "the voice" :) |
| 23:19.52 | iDunno | it's linked from digiums site. |
| 23:21.25 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:21.36 | Dr-Linux | justinu: please-enter-your.gsm .. i don't know what it says |
| 23:21.46 | justinu | play it |
| 23:25.35 | malverian[work] | [hC], Double free == stack corruption == no backtrace |
| 23:26.15 | [hC] | D'yoh |
| 23:26.29 | *** join/#asterisk FuLg0r3 (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 23:27.32 | jalsot | hi |
| 23:28.25 | justinu | malverian[work]: which softphone is causing the trouble? |
| 23:28.44 | jalsot | does anybody use speex compiled with SSE? I'm getting segmentation fault |
| 23:28.51 | jalsot | probably I'm doing something wrong... |
| 23:29.04 | malverian[work] | justinu, One we created inhouse. |
| 23:29.10 | malverian[work] | justinu, I can provide a binary however. |
| 23:29.24 | malverian[work] | justinu, It's for win32 only, but I'm able to reproduce the crash with wine. |
| 23:29.51 | justinu | if you're getting a crash on destroying call id, it must be trying to free a null pointer or something |
| 23:30.33 | justinu | can you paste the output from sip debug? |
| 23:31.56 | *** join/#asterisk linlin (i=linlin@71.194.70.5) |
| 23:32.02 | *** join/#asterisk J4k3 (i=j4k3@166.153.98.5) |
| 23:32.41 | *** join/#asterisk Mo (i=dark@g-unit.ca) |
| 23:32.47 | [hC] | does anyone have any ideas on solving double-ringing issues when the 'r' argument is NOT being passed to Dial()? |
| 23:33.02 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 23:33.13 | justinu | hc, could probably work that out also |
| 23:33.44 | [hC] | It seems to only happen when the calls go out the PRI.. as in, asterisk is generating ring, and so is the pri... strangely enough it only seems to happen on certain phones as well. |
| 23:33.59 | [hC] | I do. |
| 23:34.04 | justinu | what's the other side of the call? |
| 23:34.05 | justinu | sip? |
| 23:34.14 | P4C0 | hello guys one question, if the person in one extension is talking with the outside world, and another person (inside) dial his/her extension it gets a busy tone, isn't it possible to pass the call? so the person can put on hold the outside call and pickup the inside call then unhold the outside call?? |
| 23:34.19 | [hC] | SIP -> IAX -> PSTN PRI |
| 23:34.40 | NewSole | no [hc] this is a possible investor that came to us |
| 23:34.41 | justinu | SIP -> * -> IAX -> * -> PRI? |
| 23:34.49 | [hC] | P4C0: if the phone you are dialing supports call waiting, yes. |
| 23:34.54 | [hC] | justinu yep. |
| 23:35.09 | P4C0 | [hC], so it's a client stuff not asterisk? |
| 23:35.16 | justinu | hmm... IAX is certainly not my strong point |
| 23:35.18 | [hC] | P4C0: you can do it even if it doesnt, i suppose. |
| 23:35.27 | [hC] | i dont think its iax. |
| 23:35.50 | [hC] | Its either that under certain circumstances, * doesnt see proper call progress, |
| 23:35.51 | P4C0 | it supposed to support it, but I'm not sure if maybe i did something to prevent it... |
| 23:35.54 | justinu | call comes in on PRI, and you forward to SIP phone, causing double ring? |
| 23:35.57 | [hC] | or i simply have to turn off ringing on the PRI |
| 23:36.03 | [hC] | no, dial out from SIP phone to PRI |
| 23:36.11 | justinu | ok |
| 23:36.29 | [hC] | I should check, its possible that it happens on every call, not just pri calls... but i dont think so. |
| 23:36.30 | De_Mon | in sip.conf if no username is defined is the [section heading] used instead? |
| 23:36.33 | justinu | how about an iax2 debug trace of the outbound call? |
| 23:36.41 | [hC] | De_Mon: yes. |
| 23:37.06 | [hC] | justinu: I could probably grab one of those. have to do it tomorrow though unfortunately |
| 23:37.09 | De_Mon | [hC] if their different, I should be able to register with the username right...? |
| 23:37.29 | [hC] | De_Mon: the username= takes precedence i believe, yes. |
| 23:37.49 | justinu | hC, when you say turn ringing off on the pri, what do you mean? |
| 23:38.03 | [hC] | justinu: make the PRI not generate ringing tones when you place a call over it. |
| 23:38.09 | justinu | oh |
| 23:38.09 | [hC] | in zaptel, or something. |
| 23:38.19 | [hC] | not even sure if thats viable. |
| 23:38.25 | [hC] | just a thought. |
| 23:38.31 | justinu | IME, the far end CO always provides ringback tone on all outbound calls |
| 23:38.42 | justinu | so perhaps you can turn off the ring generator in *? |
| 23:38.46 | slappingt | does anyone have a sipura 3000 connected to several copper wire phones? |
| 23:39.01 | [hC] | well, its supposed to determine if it needs ringing on its own, unless you specify 'r' to force it |
| 23:39.06 | |omni| | only double ring out the Zap channel and not in? |
| 23:39.11 | [hC] | which leads me to believe the progress detection is not working |
| 23:39.25 | justinu | hC: i know a thing or two about pri (q931) |
| 23:39.31 | [hC] | |omni|: I presume, my inbound calls get answered before i even hear a ring |
| 23:39.38 | justinu | so theoretically with the right info, it's solvable |
| 23:39.41 | |omni| | oh, asterisk is answering |
| 23:39.49 | [hC] | yup. |
| 23:40.17 | [hC] | It doesnt seem to happen on my 7960, yet it does on my linksys spa-941 |
| 23:40.17 | |omni| | bounce to an internal (not zap) extension for a minute and see if you get the same |
| 23:40.28 | De_Mon | [hC] as far as I can tell [section name] takes president on v1.2.1 |
| 23:40.35 | [hC] | |omni|: you mean dial a locally connected extension? |
| 23:40.45 | [hC] | sip -> sip |
| 23:41.01 | jalsot | anybody experiencing asterisk crash on loading speex compiled with SSE? |
| 23:41.03 | [hC] | that does not generate double-ring, no. even when i dial another server via IAX |
| 23:41.07 | |omni| | yea, bouce one if your incoming Zap channels to one of your internal extensions on the bad phone and see if you get double ring |
| 23:41.19 | [hC] | ah i see. |
| 23:41.20 | *** join/#asterisk Thazza (n=me@203.80.44.200) |
| 23:41.23 | [hC] | I could try that |
| 23:41.37 | *** part/#asterisk viperdude (n=viperdud@84-45-168-57.no-dns-yet.enta.net) |
| 23:41.43 | [hC] | If i do get double ring, what does that indicate? |
| 23:41.55 | [hC] | I wont be able to test it from where i am at the moment :/ |
| 23:41.59 | justinu | it says that both * and the far end CO are generating ring tone on the same channel |
| 23:42.03 | |omni| | not possible that it's the ringtone on the phone? |
| 23:42.17 | *** join/#asterisk J4k3_ (i=j4k3@166.145.106.111) |
| 23:42.20 | asterisk99 | Are there any Asterisk source code gurus here? :) |
| 23:42.37 | [hC] | |omni|: its quite possible, i suppose, if the phone generates its own ring. Strange though that it doesnt happen on sip->*->sip calls, just anything that terminates on pri |
| 23:42.40 | *** join/#asterisk veepster (n=veepster@68.50.103.229) |
| 23:42.51 | |omni| | I know I can set two separate tones on two different 7960s and get a similar result, when asterisk is only ringing once |
| 23:43.13 | |omni| | ya..weird.. just ideas |
| 23:47.22 | `lyme | can asterisk anwser calls from a direct line (no dail tone, you pickup and it just dails straight out to the other party) as well as initialize connections on this same line |
| 23:47.22 | `lyme | ? |
| 23:48.08 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
| 23:48.56 | *** join/#asterisk _cleric_ (n=dacleric@p5482BD15.dip0.t-ipconnect.de) |
| 23:49.27 | ejr | Hi all. I just updated to 1.2, and I'm seeing some strange behaviour! I'm getting an intermittent "runaway asterisk process". When this happens I see the processor and memory consumption max out. |
| 23:50.02 | ejr | I just saw it when a IAX call was leaving a voicemail message..... |
| 23:50.15 | *** part/#asterisk twisty7867 (n=twisty78@adsl-gte-la-216-86-203-111.mminternet.com) |
| 23:50.27 | Ayano | Okay, I have the incoming fax working, and I'm trying to change the from address on the e-mail that it sends, does anyone know how? |
| 23:54.55 | *** join/#asterisk Pegger (n=peg@pool-68-163-192-85.bos.east.verizon.net) |
| 23:58.17 | mishehu | Ayano: depends on what you're using to receive the fax. I use a custom php script, and we just pass the script the email address... |
| 23:58.35 | Pegger | when I restart asterisk should I see upd traffic going across the wire |
| 23:58.49 | *** join/#asterisk jhelm (n=jhelm@66-128-109-118.static.stls.mo.charter.com) |
| 23:59.11 | ravenpi | You can change the sender's e-mail address a bunch of ways, depending on how you're sending the mail. With Exim, for example, you can use (as root or the exim user) the "-f" flag to set the sender. Or you can write a script. Etc. |
| 23:59.21 | jhelm | i was wondering if someone could help me with a problem with my uniden uip200's |
| 23:59.27 | justinu | pegger: what happens when you say "iax2 debug" at the CLI? |
| 23:59.35 | jhelm | they are registering with the asterisk server, but are not able to call anything |
