| 00:00.59 | ruben23 | seanbright: good thing its ok now i just deleted the app_amd.c on the asterisk dir..actually i try to install it-its a Answering Machine Detection |
| 00:01.41 | seanbright | ah. you might be using an incompatible version. |
| 00:03.25 | ruben23 | seanbright:you know how to install doxygen? for my progdocs to run... |
| 00:04.01 | seanbright | ruben23: yum search doxygen |
| 00:04.09 | seanbright | ruben23: on centos, yum is your friend |
| 00:04.11 | seanbright | learn it |
| 00:05.54 | ruben23 | ok ill do some readings.. |
| 00:06.04 | seanbright | ~thebook |
| 00:06.04 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:06.11 | seanbright | ruben23: ^^^^ that's a good place to start :) |
| 00:07.18 | jaytee | ooooh! eye candy!!!! http://www.etoday.ru/2008/11/female-bodybuilders-martin-schoeller.php?diggtoolbar ;-) |
| 00:07.31 | ruben23 | yeah i got that book.. |
| 00:07.33 | seanbright | yiiiiiiiiiiiiiked |
| 00:07.38 | seanbright | s/ed/es/ |
| 00:09.44 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 00:16.38 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 00:17.14 | ruben23 | seanbright: have question...mostly during call times...if time that we had bad calls like cant hear choppy...my head IT restart our asterisk boxes by full restart-the whole systaem restart... |
| 00:17.38 | seanbright | yikes... ok... |
| 00:17.53 | ruben23 | is there anyway that i can restart my asterisk without shutting up the system.. |
| 00:17.58 | seanbright | yes |
| 00:18.05 | seanbright | well... two things |
| 00:18.19 | ruben23 | :)it a great relief for me.. |
| 00:18.19 | seanbright | 1) you should try to resolve whatever problem you are having that is causing the choppiness in the first place |
| 00:18.28 | seanbright | 2) service asterisk restart |
| 00:18.38 | seanbright | the latter will restart asterisk itself, and not the entire machine |
| 00:19.08 | ruben23 | hmmm...i think its with our connection...but still im confuse.... |
| 00:19.24 | ruben23 | how to start the isolation process. |
| 00:19.56 | ruben23 | this problem is been a year on the run...:-D |
| 00:20.07 | Yourname` | Help needed, SIP no route to host error out of nowhere.. http://pastebin.ca/1274285 -> Guys, seriously, this is out of nowhere for no reason.. system works fine for a bit, and then suddenly no route to host starts coming up. |
| 00:20.28 | seanbright | ruben23: and how often are you bouncing the machine? |
| 00:20.49 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-183.phlapa.east.verizon.net) |
| 00:20.53 | seanbright | Yourname`: looks like an intermittent network problem |
| 00:20.58 | ruben23 | what you mean bouncing? |
| 00:21.02 | seanbright | ruben23: restarting |
| 00:21.26 | ruben23 | ok....mostly when the line are not good.. |
| 00:21.32 | ruben23 | we restart |
| 00:21.33 | seanbright | ruben23: and how often is that? |
| 00:21.37 | seanbright | once a week? once a month? |
| 00:21.43 | ruben23 | no.. |
| 00:21.45 | ruben23 | daily |
| 00:21.49 | seanbright | ughh |
| 00:21.52 | ruben23 | 4 times i guess |
| 00:21.55 | seanbright | wow |
| 00:22.00 | seanbright | is this a call center? |
| 00:22.02 | ruben23 | it worst it think |
| 00:22.07 | ruben23 | yeah... |
| 00:22.12 | seanbright | what version of asterisk? |
| 00:22.13 | ruben23 | call center outbound |
| 00:22.24 | ruben23 | ill check now.. |
| 00:22.25 | jaytee | damn! I went over 6 months without a reboot last time |
| 00:23.33 | ruben23 | its asterisk 1.2.24 |
| 00:23.57 | ruben23 | this really my headaches for ayear.. |
| 00:24.02 | seanbright | ruben23: ah... 1.2... |
| 00:24.12 | seanbright | ruben23: upgrading to 1.4 might be a good start. |
| 00:24.23 | ruben23 | it like im already used to this problem |
| 00:25.17 | ruben23 | upgrading wont harm the conf of the system...? |
| 00:25.18 | Yourname` | seanbright: Doesn't go away tho.. last time it happened, I rebooted and it worked fine |
| 00:25.53 | seanbright | ruben23: it most likely won't work with your current configuration, you'll need to do some testing before putting it into production |
| 00:26.06 | seanbright | Yourname`: does it go away on it's own if you ignore it? |
| 00:26.33 | Yourname` | seanbright: Nope |
| 00:26.36 | ruben23 | yeah its a risk...my managers might kill me... |
| 00:26.54 | seanbright | ruben23: well like i said, don't just do it. put it on another machine, build it out, and test it first. |
| 00:28.29 | ruben23 | yeah....im on practice now doing it on VM...just to familiarize. |
| 00:30.16 | ruben23 | this IRC really is a big help...for neo.. |
| 00:36.34 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 00:40.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:45.08 | *** join/#asterisk tuxfoo2 (n=tmmarini@pool-72-65-144-45.chrlwv.east.verizon.net) |
| 00:45.50 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:47.21 | *** join/#asterisk talntid (n=eric@c-67-185-179-75.hsd1.wa.comcast.net) |
| 01:02.46 | *** join/#asterisk morex (n=m@5ac648c8.bb.sky.com) |
| 01:04.31 | morex | Hey folks |
| 01:04.39 | morex | Got a problem with the DAHDI channel |
| 01:04.53 | morex | I can't log dynamic DAHDI queue members out of the queue |
| 01:05.05 | morex | Can anyone help? |
| 01:05.26 | morex | I'm using 'queue add member DAHDI/r2/0008' to add |
| 01:05.51 | morex | but 'queue remove member DAHDI/r2/0008 from queue XXXX' responds with: |
| 01:06.24 | morex | Unable to remove interface 'DAHDI/r2/2008' from queue 'CustServ': Not there |
| 01:07.10 | FruitBasket | check out "queue show" and use the tab key to complete channels. |
| 01:07.48 | morex | queue show says it's still there |
| 01:08.15 | FruitBasket | does the remove line auto complete? |
| 01:08.21 | morex | Kind of |
| 01:08.31 | morex | It just shows DAHDI/r2 |
| 01:08.35 | morex | but not the last bit |
| 01:08.39 | FruitBasket | type another / and hit tab again |
| 01:10.04 | morex | Hmm now it's working |
| 01:10.13 | morex | I'm gonna try it with the AddQueueMember manager command |
| 01:10.31 | morex | P.S. THANK YOU for helping! |
| 01:10.43 | FruitBasket | if it didn't work typing it but did work with auto complete, you're typing something incorrectly.. |
| 01:10.51 | FruitBasket | it's possible you're not typing the full channel name |
| 01:12.10 | *** join/#asterisk chendy (n=chatzill@58.60.30.239) |
| 01:12.15 | morex | Huh that's weird it is working now |
| 01:12.23 | morex | Oh well, sorry to have bothered you |
| 01:16.09 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d340944178585751) |
| 01:16.13 | *** join/#asterisk bobnormal (n=irc@221.213.47.10) |
| 01:16.23 | morex | OK the problem only strikes once a call has been delivered |
| 01:16.26 | bobnormal | hey i'm getting an error "No translator path exists for channel type Zap (native 76) to 1024" |
| 01:16.30 | bobnormal | anyone know how to fix it? |
| 01:16.34 | morex | Before, queue show gives |
| 01:16.40 | bobnormal | it occurs when trying to dial out from kphone to PSTN |
| 01:16.57 | bobnormal | and the call fails with 'service unavailable' |
| 01:17.17 | bobnormal | [Dec 1 21:00:51] WARNING[9659]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) |
| 01:17.49 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:18.04 | bobnormal | i'm pretty sure it works for other clients (hylafax via iaxmodem) though, and incoming works fine. i'm confused. |
| 01:23.53 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 01:24.36 | bobnormal | going to try recompiling .. seems its a missing codec translation driver problem |
| 01:31.05 | x86 | grrrr... this sucks |
| 01:31.18 | x86 | 1.6.0.1 on downloads.digium.com is a 404 error now |
| 01:31.22 | x86 | and 1.6.0.2 wont compile |
| 01:31.37 | FruitBasket | go to the download directory.. check the dir listing. |
| 01:33.13 | x86 | http://ftp.digium.com/pub/asterisk/ |
| 01:33.22 | x86 | go there, click on asterisk-1.6.0.1.tar.gz |
| 01:33.25 | x86 | 404 |
| 01:33.41 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 01:35.36 | FruitBasket | http://downloads.digium.com/pub/asterisk/ -- you're right. hmm. |
| 01:36.20 | FruitBasket | http://downloads.digium.com/pub/asterisk/releases/asterisk-1.6.0.1.tar.gz |
| 01:36.45 | x86 | ah ok cool |
| 01:36.58 | x86 | thanks |
| 01:37.06 | x86 | wonder why the other one no workie |
| 01:39.11 | beek | x86: because it's reported to have compilation errors. 1.6.0.3 should be available soon. |
| 01:39.25 | beek | x86: I assume that they pulled it. |
| 01:39.51 | x86 | 1.6.0.1 is the only one that compiles for me (which is 404'd), while 1.6.0.2 doesn't compile (which downloads fine) |
| 01:39.57 | x86 | seems opposite from what you're saying |
| 01:40.27 | morex | Hmm just called Digium support |
| 01:40.36 | morex | Looks like a bug in Queue with Dahdi |
| 01:40.42 | beek | x86: The "current" link is probably pointing to the 1.6.0.2 release. |
| 01:40.52 | *** join/#asterisk etfonhomey (n=chatzill@mobile-166-214-229-014.mycingular.net) |
| 01:41.56 | x86 | beek: (which works fine) |
| 01:42.07 | x86 | beek: 1.6.0.1 is the broken link |
| 01:43.34 | beek | x86: Okay... I give up. I thought I had an explanation... |
| 01:44.35 | x86 | hehe |
| 01:48.15 | *** join/#asterisk l8router (n=l8router@d122-109-92-165.sbr12.nsw.optusnet.com.au) |
| 01:50.47 | *** join/#asterisk etfonhomey_ (n=chatzill@74-143-196-254.static.insightbb.com) |
| 01:52.19 | *** join/#asterisk lkthomas (n=lkthomas@218.189.198.146) |
| 01:52.22 | lkthomas | hey guys |
| 01:52.33 | lkthomas | does anyone could explain what is DID line compare with normal phone line ? |
| 01:53.20 | jql | not all phone lines have phone numbers which cause them to ring; DID lines do |
| 01:53.35 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:54.03 | trnzmeta | DID are lines from the telco |
| 01:54.04 | trnzmeta | ? |
| 01:54.22 | jql | presumably |
| 01:54.27 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 01:57.08 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 01:57.28 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 02:03.20 | bobnormal | ok i've recompiled with latest asterisk 1.4 and asterisk-addons, but still no luck. SIP client (kphone) registers fine but gets "service unavailable" and server generates "channel.c:3035 ast_request: No translator path exists for channel type Zap (native 76) to 1024" when attempting to make a call |
| 02:03.35 | *** join/#asterisk Akiyuki (i=TuxGuy@cpe-065-184-194-136.ec.res.rr.com) |
| 02:03.48 | bobnormal | i believe i am missing ilbc (wont compile for some reason), but kphone support gsm and other codecs too. |
| 02:05.51 | RypPn | there's a script in the tarball to fetch the ilbc code if its a newer version of asterisk 1.4, post 1.4.18 or thereabouts :) |
| 02:07.02 | bobnormal | ok cheers |
| 02:07.26 | bobnormal | script name/location? |
| 02:07.41 | RypPn | under contribs iirc |
| 02:07.54 | *** join/#asterisk zmitya (n=mitya@dsl5400B0C3.pool.t-online.hu) |
| 02:07.59 | zmitya | hi all |
| 02:08.46 | zmitya | guys, if I have extensiont like this: 101, 102, 103, 104, ... etc... how can I do a simple dialplan ofr them ? They only need to call each other |
| 02:09.04 | zmitya | I tried: exten => _XXX.,s,Dial(SIP/${EXTEN}) |
| 02:09.10 | zmitya | I tried: exten => _XXX.,1,Dial(SIP/${EXTEN}) |
| 02:09.24 | zmitya | it always says: Call from '103' to extension '101' rejected because extension not found. |
| 02:09.31 | [TK]D-Fender | zmitya: the 2nd is ok, the first isn't |
| 02:09.32 | Akiyuki | Does exten 101 exist? |
| 02:09.43 | zmitya | yes, it does exists |
| 02:09.47 | [TK]D-Fender | zmitya: You need to make sure those are in the right CONTEXT <- |
| 02:10.10 | zmitya | if I say: exten => 101,1,Dial(SIP/101) it works |
| 02:10.50 | zmitya | [TK]D-Fender: ther are in the right context |
| 02:10.52 | [TK]D-Fender | zmitya: And actually, no, that pattern ISN'T good |
| 02:11.05 | [TK]D-Fender | zmitya>I tried: exten => _XXX.,s,Dial(SIP/${EXTEN}) |
| 02:11.26 | [TK]D-Fender | zmitya: that matches 3 DIGIT plus 1 or more chars of any kind. |
| 02:11.26 | zmitya | [TK]D-Fender: ok, I tried both of them |
| 02:11.38 | [TK]D-Fender | zmitya: thus a minimum length of *4* |
| 02:11.49 | [TK]D-Fender | zmitya: the "." is whats wrong |
| 02:11.52 | zmitya | [TK]D-Fender: the doc says that "." matches the zero string as well |
| 02:11.58 | zmitya | but I'm trying |
| 02:12.02 | [TK]D-Fender | zmitya: O RLY? Which? |
| 02:12.15 | zmitya | wow |
| 02:12.20 | zmitya | works |
| 02:12.26 | [TK]D-Fender | zmitya: Imagine that! |
| 02:12.28 | zmitya | in the example in sip.conf |
| 02:12.30 | [TK]D-Fender | \o/ |
| 02:12.44 | [TK]D-Fender | zmitya: Do they say that pattern is for 3 digits? |
| 02:12.48 | zmitya | ; . - wildcard, matches anything remaining (e.g. _9011. matches |
| 02:12.49 | zmitya | ; anything starting with 9011 excluding 9011 itself) |
| 02:13.07 | [TK]D-Fender | zmitya: "excluding" <- |
| 02:13.14 | zmitya | pfffffffffffffffff |
| 02:13.21 | zmitya | thanks |
| 02:13.22 | [TK]D-Fender | zmitya: and "starting" . Strats with... doesn't end with... |
| 02:13.30 | [TK]D-Fender | zmitya: Dang pesky grammar! |
| 02:13.45 | [TK]D-Fender | Thesaurus : The most literate dinosaur |
| 02:14.00 | zmitya | [TK]D-Fender: thanks again |
| 02:14.05 | zmitya | i should sleep |
| 02:14.10 | [TK]D-Fender | zmitya: Quite welcome |
| 02:24.49 | jaytee | wanna know what's scary? scary is when your boss forces you to train an airhead with ADHD on Asterisk. |
| 02:28.06 | [TK]D-Fender | jaytee: ADHD can be channeled.... airhead... well there's just no helping a severe case of acute cranial oxygen deprivation syndrome... |
| 02:29.08 | jaytee | [TK]D-Fender, I fear for the future of this company's voice infrastructure between my boss and this guy it hasn't a chance |
| 02:29.28 | [TK]D-Fender | jaytee: Is he your replacement? |
| 02:29.53 | jaytee | not hardly, he can barely troubleshoot his way out of a paper bag and my boss knows that. |
| 02:30.24 | [TK]D-Fender | jaytee: What influence will this person have over your deployment? |
| 02:30.33 | jaytee | [TK]D-Fender, none |
| 02:30.48 | jaytee | he's just there as backup in case I croak |
| 02:30.51 | [TK]D-Fender | jaytee: So he has no say and you're still in charge... wheres the "damage" then? |
| 02:31.37 | jaytee | [TK]D-Fender, if I have to delegate tasks to him that I'd feel comfortable delegating to a chimpanzee then I'm still gonna worry even though he doesn't walk on his knuckles. |
| 02:33.07 | [TK]D-Fender | jaytee: jaytee You talking about delegating to chimps requires context ;) |
| 02:33.51 | jaytee | I've been working exclusively on VOIP until this week when my boss told me I had to start taking back desktop support stuff from him because he wasn't getting things done. He had one problem with 3 computers involving opening documents through our Sharepoint portal in IE6. He struggled with this problem for 3 days. I fixed it in less than 2 hours. |
| 02:34.21 | drmessano | Did he reboot? |
| 02:34.28 | Akiyuki | rimshots |
| 02:34.56 | jaytee | lol, all the problem needed to fix it was an Office repair and applying all the recent Office updates for 2003 including SP3. |
| 02:35.10 | drmessano | Wait |
| 02:35.17 | drmessano | Theres service packs for Office 2003 |
| 02:35.19 | drmessano | Oh shit |
| 02:35.22 | drmessano | BBL |
| 02:35.24 | jaytee | hahahaha |
| 02:35.24 | [TK]D-Fender | jaytee: You're right.... I think the chimp could do it too ;) |
| 02:35.44 | [TK]D-Fender | offers up a banana |
| 02:35.52 | kb3ien | anyreason fromuser in sip.conf would override set(CALLERID(num),) ? |
| 02:35.59 | drmessano | Guys like that are what make me embaressed sometimes to get compliments from customers |
| 02:36.03 | [TK]D-Fender | kb3ien: Yes |
| 02:36.15 | [TK]D-Fender | kb3ien: thats EXACTLY what it does |
| 02:36.19 | drmessano | When I feel like I brought their business back from the bring of disaster: yay |
| 02:36.29 | kb3ien | good to know. fromuser is set to the customer string, not a numeric value, btw. |
| 02:36.31 | jaytee | he and our server admin are both hilarious. the server admin is the kinda guy who'd fight ya to the death over the last Twinkie in the box if ya know what I mean. A tortoise could outrun him. |
| 02:36.38 | drmessano | When I changed out a toner cartridge and got the same response: ^_^ |
| 02:37.10 | kb3ien | windows users are like chimps. chimps can throw their own feces without any help... |
| 02:37.12 | jaytee | And it's taken him almost 5 months to deploy MS MOM (Microsoft Operations Manager) to replace Nagios (which I had to teach him and he didn't want to learn it) |
| 02:37.14 | kb3ien | arnt like |
| 02:37.24 | drmessano | Oh here we go |
| 02:37.25 | kb3ien | buggerd up my own joke. *nuts to me* |
| 02:37.30 | [TK]D-Fender | drmessano: The people in my office are largely too afraid to do so much as change a toner cartridge. |
| 02:37.30 | drmessano | Like Linux users arent morons too |
| 02:37.58 | drmessano | Putting Ubuntu in front of them doesn't make their poo less sticky |
| 02:38.37 | kb3ien | true but i dont need to hold their hands as much (and its a good think with all that Dark Matter around). |
| 02:38.46 | drmessano | ..... |
| 02:38.48 | jaytee | brb, time to fold the laundry and put the "permanent press" on hangers so it doesn't wrinkle |
| 02:38.54 | drmessano | Right. |
| 02:39.13 | drmessano | Until they want to know where Microsoft Word is.. |
| 02:39.36 | *** join/#asterisk etfonhomey (n=chatzill@32.179.6.65) |
| 02:40.59 | kb3ien | what the difference between fromuser and defaultuser in sip.conf then? |
| 02:45.06 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 02:45.23 | kb3ien | hrm well somethings breaking my callerid, i think its upsteam, but i'll figure it out. |
| 02:45.33 | [TK]D-Fender | kb3ien: Where do you see "defaultuser"? |
| 02:47.06 | *** join/#asterisk theskinfox (n=anaxagor@ool-457c5d2e.dyn.optonline.net) |
| 02:48.06 | jaytee | fromuser modifies the way the contact info in the header is structured when you send a SIP INVITE. I have no clue what defaultuser is. |
| 02:48.40 | [TK]D-Fender | jaytee: Neither does the WIKi, and I'm trying to find something that tells me it isn't another made-up parm |
| 02:48.52 | drmessano | Never heard of it |
| 02:48.52 | kb3ien | was in sip.conf sample file from, somewhere... |
| 02:49.00 | drmessano | madeup.com |
| 02:49.02 | kb3ien | hrm nice. i dint annotate the src. |
| 02:49.06 | kb3ien | must be. |
| 02:49.22 | jaytee | maybe it's a new parm in 1.6.2. From what I hear in that version all SIP accounts are peers and SIP has no friends or users anymore |
| 02:49.26 | [TK]D-Fender | kb3ien: So from the distributed sample file? |
| 02:49.29 | kb3ien | so its the normal way to send your account name to your upstream provider? |
| 02:49.32 | [TK]D-Fender | kb3ien: What version? |
| 02:49.48 | [TK]D-Fender | kb3ien: set the callerid |
| 02:50.00 | [TK]D-Fender | kb3ien: and standard auth does its own deal |
| 02:50.13 | kb3ien | okay, thats what i was expecting. |
| 02:50.14 | [TK]D-Fender | kb3ien: Other helpful settings "sendrpid=yes" |
| 02:50.31 | beek | Hey guys... it it normal for a PRI to drop occasionally? I've been fighting this damned thing now for over a week. I keep getting: PRI got event: Alarm (4) on Primary D-channel of span 1 at random intervals. |
| 02:50.42 | jaytee | maybe he confused defaultuser with username parm |
| 02:50.58 | jaytee | beek, that's not normal |
| 02:51.32 | beek | jaytee: Thanks. I have been dealing with Level 3 Communications and the only tech support I get there is some guy in India. |
| 02:51.40 | beek | Trying to track this down is a major PITA. |
| 02:51.55 | jaytee | who's probably huddling under a desk afraid to come out |
| 02:52.12 | beek | No doubt. |
| 02:52.22 | *** join/#asterisk jtodd (n=jtodd@90.sub-70-221-185.myvzw.com) |
| 02:52.45 | beek | This is a Sangoma a104d card, dahdi and asterisk 1.6.0.1. |
| 02:53.27 | beek | Is there any traces or anything that I can take to figure out WTF is going on? |
| 02:53.39 | jaytee | hmmm, haven't run my PRI's with DAHDI yet. still using Zaptel but I use Digium TE212P cards with real HWEC instead of that fake software shit |
| 02:53.58 | beek | This card has the HWEC, too. |
| 02:54.39 | jaytee | I'd bet it's the distant end of your PRI |
| 02:54.53 | jaytee | how long does it last? |
| 02:54.59 | beek | The one that terminates in India? ;-) |
| 02:55.10 | beek | A couple of seconds. |
| 02:55.20 | jaytee | serious? you've got a PRI going all the way to India? |
| 02:55.29 | jaytee | that's nuts |
| 02:55.53 | beek | It seems like it, given that is where the support is coming from. Actually, this goes to the CO about two miles away. |
| 02:56.43 | beek | wanpipemon -i w1g1 -c Ta says I have two out-of-frame errors since I restarted everything at 2:00 p.m. |
| 02:56.47 | jaytee | what do you have for CSU's? |
| 02:57.32 | beek | Jeez... it's been a long day. Let me go look... BRB |
| 02:59.40 | beek | jaytee: They're Adtran T1 ESF CSU ACE |
| 03:00.22 | jaytee | beek, that's what I've got |
| 03:00.59 | jaytee | mine have alarm led indicators on them. |
| 03:01.11 | beek | jaytee: Ditto. |
| 03:01.48 | jaytee | beek, how many spans total in your PRI? |
| 03:02.07 | beek | One span to the PRI. Another to our legacy PBX. The third to a channel bank. |
| 03:02.22 | beek | Span 1 (to the PRI) is set for B8ZS ESF normal timing. |
| 03:02.25 | jaytee | and you're only getting alarms on the PRI span? |
| 03:02.29 | beek | Yes |
| 03:02.35 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:03.24 | beek | Spans 2 and 3 are B8ZS ESF master timing using Span 1 as the source. |
| 03:03.33 | jaytee | I haven't actually ever seen a PRI that doesn't use B8ZS ESF. I've seen fractional T1 using AMI but that was back in 94 |
| 03:04.07 | jaytee | sounds like your card is working ok then. The frame slips are probably coming from their end. |
| 03:04.25 | sprite-- | Hey! I am doing a site similar to niteflirt.com. I need to be able to call 2 parties, bridge the connections, but keep them in seperate contexts. I am planning on doing the webpage in RoR and it looks like I can easily intergrate it with adhearsion. Does the Asterisk 1.6 bridge support what I need to do? If not what is the best way to do it? From what I read MeetMe conference is not very performance effective. What is the best way to handle a large number |
| 03:04.41 | jaytee | but Span 1 gets it's timing from your telco, correct? because it should. |
| 03:05.04 | beek | Correct. If I pastebin the config would you mind taking a peek? |
| 03:05.34 | jaytee | sprite, for two party calls use app_bridge |
| 03:06.40 | jaytee | sprite, you might want to look at the AJAM libs to use with RoR instead of Adhearsion |
| 03:08.10 | beek | jaytee: http://www.pastebin.ca/1274524 |
| 03:08.11 | jaytee | although Adhearsion is more Ruby oriented |
| 03:08.39 | sprite-- | jaytee: Thanks. Where is there a list of Asterisk applications such as Bridge that I can download and install? |
| 03:09.15 | sprite-- | Going to give Gentoo + Asterisk 1.6 a shot. Been messing with the AsteriskNow 1.5b for a few days and I really like Asterisk so far. |
| 03:09.17 | jaytee | sprite, it's only in 1.6 and you can find it in the /docs in the tarball |
| 03:09.58 | sprite-- | Ahh ok. Does Bridge let me keep the channels in a seperate context so I can unbridge them if I want to? |
| 03:11.28 | beek | jaytee: according to the docs on Sangoma's site things are set appropriately. |
| 03:11.55 | *** join/#asterisk zchaos (i=none@CPE001aa0829288-CM001ade84db36.cpe.net.cable.rogers.com) |
| 03:14.47 | jaytee | beek, looks fine in your system.conf |
| 03:15.33 | beek | jaytee: Thanks... I appreciate another set of eyes on it. I'm convinced that its the telco but getting those bastards to actually do something about it is a challenge beyond compare. |
| 03:16.06 | jaytee | beek, yep that's always the big problem with PRI's and dealing with the telcos. |
| 03:17.01 | beek | jaytee: While we were TelCove's customer my support center was 45 minutes away. The service was great. Now, things suck royally and the only tech support I get is from India. My sales rep is in Oklahoma. |
| 03:17.34 | jaytee | only way to prove it is to loopback at your end and wait for a sufficient interval. If you get no more out of frame errors then it's gotta be their end. If it was your card it would be screwing with the other spans too. |
| 03:18.13 | beek | jaytee: Hmmm... This is a loop back through the CSU? |
| 03:18.29 | jaytee | Level3 is the perfect example of big not always being better. |
| 03:19.39 | jaytee | beek, you could loopback at the adtran, loopback from inside the card using dahdi_tool or you could just make a T1 crossover 1 -> 4, 2 ->5 if you have the crimps and tool and slap that on the span. |
| 03:20.32 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 03:20.55 | jaytee | my CSU has a set of DIP switches for LBO and for doing loopback on the CPE side and the NET side. |
| 03:21.06 | Micc | can I do an include in agents.conf and queues.conf? |
| 03:21.14 | jaytee | might piss off your buddies in India though :-) |
| 03:21.29 | beek | jaytee: fuck 'em. I've had it. |
| 03:21.34 | jaytee | lol |
| 03:22.25 | beek | I'm investigating the dahdi_tool to see if I can loop back that way. I also have a terminal attached to the Adtran, so perhaps I can get into the configuration and loop it back through there. |
| 03:23.07 | jaytee | I've had really good luck with Time Warner but when my PRI was with AT&T I had to borrow a friends T-Berd and pull dumps of all the frame slips coming off the circuit and then call screaming at them. The word lawyers seemed to get a response. |
| 03:23.27 | beek | That's next up. |
| 03:23.28 | jaytee | miraculously the slips stopped. |
| 03:23.51 | beek | Well, this morning my PRI dropped at 0600. I didn't get them to get it back up until 1400. |
| 03:24.14 | beek | They called and said "all was fixed" and here I am, back at 10:23, looking at another. |
| 03:24.33 | jaytee | where are you? |
| 03:24.37 | beek | Pennsylvania |
| 03:24.49 | jaytee | yeah? what part? |
| 03:24.58 | beek | SW of State College |
| 03:25.23 | jaytee | ok, know where that is vaguely. Steelers kicked the Pats butts sideways sunday. |
| 03:25.42 | beek | I'm about 2.5 hours from Pittsburgh. Where are you? |
| 03:26.02 | jaytee | I didn't feel too bad even though I'm originally from south of Boston. Now I'm living in exile in Indianapolis |
| 03:26.44 | beek | This is my first experience interfacing Asterisk to a PRI... I've learned more in the last week about PRIs then I ever thought I'd have to know. |
| 03:27.02 | beek | Where would you suggest I do the loop back? At the CSU? |
| 03:27.33 | jaytee | I used to fly into Pittsburgh and drive to Morgantown, WV or catch a puddle jumper to Williamsport, PA to do installs and upgrades at retail offices there. I liked Pittsburgh, especially the airport. |
| 03:27.55 | beek | Williamsport? Really? I grew up about 13 miles from there. |
| 03:28.02 | jaytee | beek, yeah if you can. |
| 03:28.32 | jaytee | is your office in operation now or is it after hours? |
| 03:28.36 | beek | What's a reasonable amount of time to run this test? I have the phone system until 0700. |
| 03:28.44 | beek | After hours... WAY, after hours. |
| 03:28.50 | jaytee | how long was it between alarms? |
| 03:29.08 | beek | Well, initially it started at around 8 hours. Then 6 hours, then hourly. |
| 03:29.29 | jaytee | try looping it back for an hour or so if you can afford the time |
| 03:29.42 | drmessano | Speaking of loopback.. |
| 03:29.49 | jaytee | yes? |
| 03:29.53 | drmessano | 127.0.0.1 <------ HACK ME! |
| 03:29.59 | drmessano | 127.0.0.1 <------ GO ON! |
| 03:30.04 | jaytee | hahaha |
| 03:30.05 | drmessano | 127.0.0.1 <------ U NO U WANTU |
| 03:30.14 | jaytee | u r gay! |
| 03:30.14 | drmessano | ok, done |
| 03:30.28 | jaytee | zomg! |
| 03:30.33 | drmessano | GAY HUH? WANNA FIGHT ABOUT IT -------> 127.0.0.1 |
| 03:30.43 | jaytee | lolz |
| 03:31.02 | beek | jaytee: I'm off to the wiring closet and see if I can get that loop back configured. I have a terminal attached to the CSU so I can easily configure it. |
| 03:31.05 | drmessano | JAYTEES HOUSE --------> 127.0.0.2 |
| 03:31.08 | beek | jaytee: Thanks very much. |
| 03:31.27 | jaytee | beek, good luck! sucks being in the trenches late at night. |
| 03:31.37 | beek | jaytee: Yep. |
| 03:32.03 | jaytee | beek, one more thing. |
| 03:32.33 | jaytee | you may need to reset your timing in system.conf while you test, but I'd only do it if it throws an alarm about clock sync |
| 03:33.07 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 03:33.20 | jaytee | beek, you're only using 3 of your 4 ports, right? |
| 03:34.47 | beek | jaytee: yes |
| 03:35.27 | beek | jaytee: Am I looking for test, line or local loopback? |
| 03:36.01 | jaytee | you could try setting up the unused port as a PRI_NET span supplying timing and use a crossover cable to go from your span 1 to that span (4?) |
| 03:36.07 | jaytee | local loopback |
| 03:38.04 | jaytee | whenever I test a dual port card I setup one span as PRI NET with timing and the other as PRI CPE and route calls out the context for the NET span and into the incoming context for the CPE span to test phone to phone over the circuit. |
| 03:40.49 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 03:40.51 | beek | jaytee: I see. |
| 03:42.31 | beek | jaytee: that sounds like a great idea. I think I'll engineer that tomorrow morning and start it right after the office closes tomorrow evening. That way I can have a few hours to watch it. |
| 03:42.53 | beek | jaytee: I think I'm going to head home and get some sleep. I really appreciate your help this evening and I'll let you know how I make out. |
| 03:43.29 | jaytee | beek, you can even run test calls that way if you setup some temporary matches in the contexts for the spans and then comment them out later. |
| 03:43.53 | jaytee | beek, have a good nite |
| 03:44.15 | beek | jaytee: I never thought of that before but you've given me some great ideas. I'll be ready to do battle tomorrow! |
| 03:44.16 | beek | GN |
| 03:44.46 | jaytee | sometimes you just gotta grab the bull by the tail and face the situation :-) |
| 03:45.21 | [TK]D-Fender | jaytee: And if the situation is a very angry bull with sharp horns? |
| 03:45.37 | [TK]D-Fender | RUN FORREST RUN!!!!!! |
| 03:45.40 | jaytee | lol |
| 03:45.51 | jaytee | "BATTLE!!!!" |
| 03:45.57 | [TK]D-Fender | yarrr! |
| 03:46.06 | jaytee | I love that scene in Michael |
| 03:46.14 | jaytee | the poor bull |
| 03:46.35 | [TK]D-Fender | jaytee: I love the bull in the Looney Tunes "Bully for Bugs" |
| 03:46.41 | [TK]D-Fender | jaytee: Sly Bull > all |
| 03:47.07 | jaytee | god, I haven't seen that in ages! |
| 03:47.20 | jaytee | shit, I think Reagan or Carter was President |
| 03:47.27 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.21) |
| 03:48.01 | theskinfox | so can anyone help me out with "one touch" call recording aka automon? |
| 03:48.18 | [TK]D-Fender | jaytee: http://www.youtube.com/watch?v=GEzyNNhOeBw |
| 03:48.30 | [TK]D-Fender | jaytee: @ 5:11 I took a DVD screen-shot for my ex :) |
| 03:49.26 | theskinfox | I want to be able to play 1 sound file when starting the recording, and play another sound file when stopping it |
| 03:49.27 | jaytee | awesome! I didn't know this was on YT. |
| 03:50.06 | [TK]D-Fender | theskinfox: No such option |
| 03:51.09 | [TK]D-Fender | jaytee: Do skip to the pic at least... its that leer he gives... |
| 03:51.54 | drmessano | If it had red hair, I would buy it a house |
| 03:51.56 | jaytee | hahahaha |
| 03:52.21 | theskinfox | <[TK]D-Fender>: do you know of anyway I could put in a feature request? I'm sure others could/would find it useful... I've been trying to butcher the code, but being I'm not a coder i've had no luck |
| 03:53.26 | [TK]D-Fender | theLack of skill is a near prerequisite for butchery. I'm sure you're on the right path! |
| 03:53.30 | drmessano | I guess that would explain it |
| 03:53.55 | theskinfox | rofl |
| 03:53.55 | jaytee | hahahaha |
| 03:53.56 | theskinfox | thx |
| 03:54.39 | drmessano | I spent a month trying to code a WMA audio codec for asterisk.. but I couldn't get notepad to open from the command line |
| 03:54.43 | drmessano | start notepad.exe my ass |
| 03:55.08 | jaytee | do you know about the .LOG option in notepad? |
| 03:55.09 | [TK]D-Fender | theskinfox: You can use the "bounty" page on the WIKi if you care enough |
| 03:55.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:55.24 | [TK]D-Fender | jaytee: notepad++ = awesome |
| 03:55.35 | theskinfox | notepad++ is pretty awesome |
| 03:55.37 | [TK]D-Fender | jaytee: http://notepad-plus.sourceforge.net/uk/site.htm |
| 03:55.38 | jaytee | Gvim FTW! |
| 03:56.09 | theskinfox | <[TK]D-Fender> : thanks, i'll drop in a request |
| 03:56.27 | theskinfox | jaytee : vim ftmfw |
| 03:56.32 | drmessano | nano.exe <--- |
| 03:56.36 | [TK]D-Fender | jaytee: VIM is extreme power with little grace. Notepad++ is pretty powerful with lots of grace :) |
| 03:57.16 | drmessano | Oh man |
| 03:57.20 | drmessano | pownce is shutting down |
| 03:57.21 | jaytee | I like vim or Gvim because it has the asterisk syntax checking (arguably not all that great but better than nothing) |
| 03:57.38 | jaytee | yeah, I read that |
| 03:57.52 | *** join/#asterisk mattzerah (n=matt@ozvoip.dsl.onthenet.net) |
| 03:58.18 | drmessano | pownce always invoked images of emodouche kitters blogging from their iphones with moleskin cases |
| 03:58.36 | drmessano | moleskine* |
| 03:59.36 | jaytee | Moleskine stuff is a big seller at Barnes and Noble lately |
| 04:00.09 | jaytee | all the yuppy wannabee Hemingways and Picassos want them |
| 04:00.39 | mattzerah | any devs here that may be able to answer a question or two about hints ? |
| 04:01.07 | jaytee | mattzerah, might have better luck in #asterisk-dev |
| 04:01.15 | mattzerah | ahh, cool, thanx :) |
| 04:02.37 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:03.46 | sprite-- | 1.6.0.2 is latest stable? |
| 04:03.53 | drmessano | yes |
| 04:04.05 | drmessano | As stable as 1.6.0 is |
| 04:04.52 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:06.13 | drmessano | I am currently not able to make SIP TCP work on 1.6 and expect a refund any day now |
| 04:06.24 | theskinfox | SWEET, i got it to work, yay for butchered code |
| 04:06.37 | drmessano | yay fur notpad |
| 04:06.50 | theskinfox | vi/vim actually |
| 04:07.36 | theskinfox | so hre's a really dumb question |
| 04:07.38 | *** join/#asterisk mattzerah (n=matt@ozvoip.dsl.onthenet.net) |
| 04:07.48 | theskinfox | i'm very new to asterisk, only been messing around with it for a week |
| 04:07.58 | [TK]D-Fender | mattzerah: Just ask, maybe you'll get an answer |
| 04:08.00 | echinos | yeah, me too :) |
| 04:08.30 | [TK]D-Fender | drmessano: Don't like your free download? We'll give you DOUBLE your money back.... |
| 04:08.36 | theskinfox | what are the differences between 1.4.x and 1.6 branches? I haven't really found any "basic feature over views"... just the changelogs which go into a lot more detail than I need |
| 04:08.46 | drmessano | I actually have never used asterisk.. i'm jaytees brother, and he makes me sit here and entertain him |
| 04:08.57 | jaytee | lol |
| 04:09.08 | drmessano | jaytee: I am makin monkeyface |
| 04:09.16 | mattzerah | ok, with hints, i have a few linksys phones with blf working correctly...... |
| 04:09.27 | mattzerah | first call comes in -> extension shows riniging |
| 04:09.43 | mattzerah | first call gets answered -> extension shows inuse |
| 04:09.51 | [TK]D-Fender | theskinfox: Quite a few articles on the big stuff. Key items : SIP TCP support, more native faxing, better T.38 support including termination, TLS, devstate built in, and a bunch more. |
| 04:10.02 | mattzerah | second call comes in (first call still being connected) -> hint shows ringing&inuse |
| 04:10.06 | sprite-- | jaytee: 1.6.0.2 has the Bridge function? What doc file documents it? |
| 04:10.19 | mattzerah | first call gets put on hold, and second call gets answerd -> hint shows idle |
| 04:10.40 | jaytee | sprite, do a core show application bridge |
| 04:10.45 | [TK]D-Fender | mattzerah: pastebin a channel dump and a hint dump |
| 04:10.45 | jaytee | at the CLI |
| 04:11.02 | mattzerah | okay, hang on i'll reproduce :), thanx [TK]D-Fender |
| 04:11.03 | sprite-- | jaytee: Didn't install 1.6 yet. |
| 04:11.09 | [TK]D-Fender | ~pb |
| 04:11.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:12.52 | theskinfox | I'm very interested in the faxing and t.38 support, but One of the things i read about 1.6 is that 1.4 dialplans may no longer work? |
| 04:14.05 | [TK]D-Fender | thelittle changes as always |
| 04:14.14 | [TK]D-Fender | theforget sweeping statements like that |
| 04:14.26 | [TK]D-Fender | theskinfox: and get a more unique nick! |
| 04:15.12 | theskinfox | LOL |
| 04:15.17 | jaytee | sprite, not sure where in the docs it's documented but it should be in there. there's an api document in OpenOffice odt format and a bunch of others. |
| 04:16.28 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-147-101.telkomadsl.co.za) |
| 04:17.22 | jaytee | sprite, here's a paste of core show application bridge http://www.pastebin.ca/1274583 |
| 04:18.05 | mattzerah | [TK]D-Fender, http://pastebin.com/m13bc4331 |
| 04:18.22 | mattzerah | much appreciated that you (or anyone of course) can look at this and see whats going on ! |
| 04:19.33 | Carlos_PHX | mattzerah: I run the Linksys phones and could try a test, but we're moving the office around and they're all down. Maybe later if I choose "fixing the network" over "drink beer." |
| 04:19.43 | Carlos_PHX | mattzerah: What firmware? What model phones? |
| 04:19.45 | [TK]D-Fender | mattzerah: Yeah, SIP/400 is on 2 calls |
| 04:19.53 | sprite-- | jaytee: thanks |
| 04:20.11 | [TK]D-Fender | mattzerah: Looks like a bug to me... |
| 04:20.16 | lkthomas | jql: you there ? |
| 04:20.18 | mattzerah | latest firmware (6.1.3a from memory) 400 = spa962 (with SPA932), others = spa942 |
| 04:20.31 | jaytee | personally I think anyone who chooses "fixing the network" over "drink beer" needs their head examined |
| 04:20.41 | mattzerah | yea, i thought as much - i might need to look into the hint code |
| 04:20.55 | lkthomas | our company got a DID line, I assume it is not same as normal phone line ? |
| 04:21.21 | xacatecas | can anybody explain the difference between SIP/1234@user and SIP/user/1234 ? |
| 04:21.29 | sprite-- | So it bridges the audio, but the users remain in their seperate dialplans? |
| 04:22.29 | jaytee | sprite, we're talking channels not contexts |
| 04:23.44 | xacatecas | jaytee, can this for example be used to "park" a user somewhere in say MOH() and to then bridge him with another caller at some stage? |
| 04:24.30 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 04:25.51 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 04:25.55 | jaytee | xacatecas, you'd have do some dialplan scripting tricks to do that but I'd say it's possible. when you say "at some stage" that's kind of vague. shit, that's totally vague. |
| 04:26.21 | xacatecas | jip. i know. but the point being that it can be done :) |
| 04:27.17 | jaytee | there's not much you can't do with Asterisk and for that there's always Viagra |
| 04:27.33 | xacatecas | crude but true. |
| 04:27.58 | Carlos_PHX | Hey, can anyone outside the US tell me if your ISDN BRI SPIDs are something other than the phone number followed by four ones? |
| 04:28.05 | Carlos_PHX | Just a curiousity question. |
| 04:28.21 | jaytee | there's a dead cat in there somewhere |
| 04:29.13 | drmessano | Oh god |
| 04:29.20 | drmessano | You said SPID's |
| 04:29.26 | drmessano | I just got the eeby jeebies |
| 04:29.28 | jaytee | yes, he did! |
| 04:29.38 | drmessano | 0101 |
| 04:29.38 | jaytee | i thought it was heeby jeebies |
| 04:29.43 | drmessano | cringes |
| 04:30.00 | drmessano | DSN's and SPIDs can KMA |
| 04:30.07 | xacatecas | o.O |
| 04:30.08 | jaytee | [TK]D-Fender, pastebin.ca now has a Firefox add-on :-) |
| 04:30.57 | drmessano | Just getting flashbacks to setting up broadcast equipment on ISDN BRI's using G.722 |
| 04:30.58 | Carlos_PHX | BRIs kick ass. Everyone should have one. |
| 04:31.01 | Carlos_PHX | wiggles bait |
| 04:31.05 | sprite-- | jaytee: Cool, seems like bridge will do exactly what I need then. Except I don't see any support to unbridge the channels? |
| 04:31.56 | jaytee | sprite, the bridge is broken when one end or the other hangs up the phone |
| 04:33.40 | sprite-- | jaytee: Yeah. I suppose I could change it myself to add the functionality to unbridge. I got have lots of experience with C/C++ just not in a linux environment. Basically I need to be able to unbridge on demand. |
| 04:34.16 | jaytee | sprite, you're better of asking about that in #asterisk-dev probably |
| 04:35.29 | xacatecas | i have another use for Bridge() ... instead of the M or U options to Dial you can split them off using G, which means I can now control the two channels separately until such time as I'm happy to Bridge them. |
| 04:35.33 | [TK]D-Fender | sprite--: "soft hangup [channel]" :) |
| 04:35.38 | [TK]D-Fender | Simplest solutions = best |
| 04:36.17 | [TK]D-Fender | sprite--: AMI Redirect does 1/2 a split |
| 04:36.41 | xacatecas | [TK]D-Fender, I suspect he wants each channel to go into a separate channel again after unbridging. |
| 04:37.11 | xacatecas | what happens if I pass multiple L() options to Dial()? |
| 04:37.30 | [TK]D-Fender | xacatecas: probably takes the alst... if you're LUCKY |
| 04:37.38 | jql | I've written a Bridge function before. it's not all that complicated -- doesn't even require digging through that much source to figure ut |
| 04:37.57 | xacatecas | [TK]D-Fender, no, if I'm really lucky it takes the lesser time limit. |
| 04:38.41 | [TK]D-Fender | xacatecas: and how / why are you passing repeat parameters to dial? |
| 04:39.07 | xacatecas | core routing puts a L() on my dialopts var to time-limit calls based on available credit. |
| 04:39.55 | xacatecas | now a client is selling "translation" time and wants to limit a call duration, since the client has credit with me which I don't want her to exceed the simple, stupid solution will end up having multiple L() options. |
| 04:40.20 | xacatecas | but I suspect I can just pass a "maxlimit" variable in to my system which will do the "lesser" thing instead of multiple L() options. |
| 04:41.52 | xacatecas | not that much more work and will do the trick. A trickier question is this though: Since the L() option limit from the point when the called party picks up, is there any way to limit from the time when the call got bridged? |
| 04:42.03 | [TK]D-Fender | xacatecas: this is your dilaplan... parse away |
| 04:42.11 | [TK]D-Fender | xacatecas: You've created your won mess. |
| 04:42.38 | xacatecas | lol no, i've just spent two weeks cleaning the garbage out. that's pretty much the only nasty that's left. |
| 04:44.05 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:44.43 | [TK]D-Fender | xacatecas: It has grown in power and anger in your negligence... |
| 04:45.34 | jaytee | parry and thrust, feint and attack |
| 04:45.38 | drmessano | Wow |
| 04:46.17 | drmessano | Michael Stipe singing It's the end of the world as we know it, without cheat sheets |
| 04:46.36 | jaytee | ??? |
| 04:47.09 | xacatecas | 66 extensions (161 priorities) in 57 contexts. <-- used to be MUCH, MUCH worse (490 extensions (619 priorities) in 560 contexts). exact same functionality. |
| 04:47.29 | drmessano | Just watching on YouTube.. recent video of him singing it without the lyrics in front of him |
| 04:47.53 | drmessano | I guess after 20+ years, some things just stick |
| 04:48.18 | xacatecas | it took him 20 years? :p |
| 04:48.41 | jaytee | I love the video for Michael Andrews performing "Mad World" from the Donny Darko theme. |
| 04:48.51 | drmessano | Just about |
| 04:49.31 | *** join/#asterisk ShaneAu (n=shane@203.56.250.52) |
| 04:49.34 | Carlos_PHX | WTF, looks like in 1.6 you can no longer do a set with two items separated by a pipe. Anyone know the replacement method? |
| 04:49.37 | drmessano | For years he used a stand in front of him |
| 04:50.45 | jaytee | Carlos_PHX, try a comma |
| 04:50.58 | Carlos_PHX | I thought they were going to pipe, not the other way. |
| 04:51.04 | jaytee | nope |
| 04:52.01 | [TK]D-Fender | xacatecas: Shit looks pretty good... when compared to crap ;) |
| 04:52.28 | jaytee | yeah, but crap usually has better "texture" |
| 04:52.36 | Carlos_PHX | Damn, that doesn't work either. |
| 04:52.53 | Carlos_PHX | exten => 16023258422,s,Set(FAXADDRESS=carlos@televolve.com,LOCALSTATIONID=TelEvolve) |
| 04:53.06 | [TK]D-Fender | Carlos_PHX: "s" is NOT a priority |
| 04:53.10 | Carlos_PHX | That just sets FAXADDRESS to be everything else on the line. |
| 04:53.16 | Carlos_PHX | Yes it is. |
| 04:53.26 | [TK]D-Fender | Carlos_PHX: Since when? |
| 04:53.31 | Carlos_PHX | 0.9x |
| 04:53.38 | Carlos_PHX | Or maybe 1.0 |
| 04:53.39 | jaytee | s is not a priority in any 1.x release |
| 04:53.42 | xacatecas | [TK]D-Fender, you're extremely friendly today. either way. I need to be off. |
| 04:53.44 | [TK]D-Fender | Carlos_PHX: No, "s" is an Asterisk Standard Extensions |
| 04:53.47 | Carlos_PHX | Certainly way before 1.2 |
| 04:54.03 | Carlos_PHX | No, really, that does work. It's a "same" priority. |
| 04:54.08 | Carlos_PHX | We use it extensively. |
| 04:54.12 | jaytee | Carlos_PHX, put the bong down and step away from the "coffee bar" |
| 04:54.25 | Carlos_PHX | EVERY one of our customer configs uses it. |
| 04:54.34 | [TK]D-Fender | Carlos_PHX: Really... I'd have to go read that somewhere... |
| 04:54.50 | [TK]D-Fender | Carlos_PHX: "same" as what? |
| 04:55.01 | Carlos_PHX | I don't know where it's documented. Kevin Fleming set up our systems originally, he may have just read it in the code. |
| 04:55.03 | jaytee | I've never seen that in any of the WIKI articles. |
| 04:55.06 | Carlos_PHX | Same as previous. |
| 04:55.11 | Carlos_PHX | I'll pastebin an example. |
| 04:55.14 | [TK]D-Fender | carWhats the poitn? |
| 04:56.07 | Carlos_PHX | http://televolve.pastebin.com/m5f25270e |
| 04:56.30 | Carlos_PHX | But anyway, that doesn't affect the multiple variable issue. |
| 04:56.38 | Carlos_PHX | Which did work at least in 1.2 |
| 04:57.29 | jaytee | wow, this is from the WIKI about parked calls: If you have a more complex dialplan and want to be able to Goto() a more elaborate 'parkedcalls' handler then you'll need to be sure to include a handler for the 'i' priority to catch calls to parkinglot without call in them as well as the 's' priority to give timeouts somewhere to go, thus: |
| 04:57.45 | Carlos_PHX | I had no idea the 's' priority was a secret. |
| 04:58.09 | jaytee | well, it's not anymore now that you've blabbed! |
| 04:58.15 | jaytee | :-) |
| 04:59.05 | [TK]D-Fender | Carlos_PHX: No, just poorly documented |
| 04:59.14 | [TK]D-Fender | Carlos_PHX: And I DO think I recall mention of this before... |
| 04:59.29 | [TK]D-Fender | Carlos_PHX: Not the full breakdown of its purpose, but attesting to its existance. |
| 04:59.49 | [TK]D-Fender | Carlos_PHX: Still... EW!!! |
| 04:59.57 | [TK]D-Fender | Carlos_PHX: Go code that stuff clean! |
| 05:07.03 | jaytee | hmmm, found an archived thread from the digium listserv group about the n and s priorities in CVS but not in release from back in 2005 posted by Kevin Fleming |
| 05:08.09 | jaytee | and it was supposed to show up in release in 1.2 but the only mention of it on the WIKI is in that one part about parked calls |
| 05:08.27 | jaytee | nothing in "the book" for sure |
| 05:08.46 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:09.07 | jaytee | well, time for bed. nite all |
| 05:09.17 | Carlos_PHX | Damn, my hackintosh crashed. |
| 05:09.38 | drmessano | You poor bastard |
| 05:09.47 | drmessano | Did you win the MAC in a raffle? |
| 05:10.16 | Carlos_PHX | I built it, that's why it's a hackintosh. |
| 05:10.29 | Carlos_PHX | Generic hardware with Mac OS |
| 05:13.57 | Carlos_PHX | So it looks like SET no longer takes multiple variables, you use ARRAY |
| 05:15.08 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 05:31.21 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 05:41.22 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:41.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:45.45 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:51.37 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 05:51.39 | [TK]D-Fender | *yawn* hcekout time |
| 05:51.42 | [TK]D-Fender | checkout* |
| 05:51.44 | [TK]D-Fender | later all |
| 06:22.56 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 06:23.28 | *** join/#asterisk saftsack (n=oliver@g226134172.adsl.alicedsl.de) |
| 06:31.55 | sprite-- | Will upgrading AsteriskNOW's installation of Asterisk from 1.4 to 1.6 break it? |
| 06:33.42 | drmessano | Its an option, is it not? |
| 06:34.25 | sprite-- | ? I was wondering if I just do a yum install asterisk16 if it will overwrite config files, etc? |
| 06:35.57 | drmessano | No |
| 06:36.46 | *** join/#asterisk ipguy (n=ipguy@124-171-250-44.dyn.iinet.net.au) |
| 06:36.58 | ipguy | hi all |
| 06:37.31 | ipguy | whats the smallest asterisk device currently on the market ? |
| 06:37.59 | drmessano | Smallest how? |
| 06:38.08 | sprite-- | drmessano: How do I upgrade? It says to install an upgrade package but I'm not sure where to find it? |
| 06:38.10 | ipguy | size ? |
| 06:38.18 | ipguy | drmessano: size |
| 06:38.35 | jql | probably mac mini. heh |
| 06:38.46 | jql | might not be entirely joking |
| 06:38.50 | drmessano | WRT54G |
| 06:39.09 | ipguy | ok with FXO FXS ports |
| 06:39.13 | jql | there ya go |
| 06:39.16 | jql | :) |
| 06:39.19 | drmessano | no |
| 06:39.29 | drmessano | You didnt specify analog ports |
| 06:39.37 | ipguy | ok, i am now. |
| 06:40.20 | ipguy | smallest imdebbed * with analog port, one will do |
| 06:40.40 | drmessano | FXS or FXO |
| 06:40.44 | drmessano | You didnt specify |
| 06:41.10 | ipguy | FXS (that will allow me to plug my phone into t right ?) |
| 06:41.35 | drmessano | Have you even used Asterisk before? |
| 06:41.49 | ipguy | drmessano: yes, setup a test box the other day. |
| 06:41.58 | drmessano | So "No" |
| 06:42.00 | ipguy | drmessano: worked perfectly |
| 06:42.17 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 06:43.07 | ipguy | no? i said i setup an * installation, configured it to connect to my VOIP provider, setup a few extentions internally... blah blah blah |
| 06:44.19 | ipguy | it works, great, now i want to use it at home but don't want to have a PC set aside for it's use. a device is a much better idea if they support full implementation of * |
| 06:44.39 | ipguy | and the device would need one FXS port |
| 06:46.57 | ipguy | drmessano: have you lost interest in the conversation ? |
| 06:49.58 | drmessano | In order to get a small device with a minimum of a single FXS port, you'll spend a disproportionate amount of money |
| 06:50.16 | drmessano | You're better off with a small PC and an ATA |
| 06:50.43 | ipguy | i was afraid you'd say that |
| 06:51.16 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 06:51.18 | ipguy | in that case i'll just dd-wrt my router and install * on it |
| 06:51.39 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 06:51.45 | tengulre | hi,all |
| 06:51.53 | ipguy | now all i need to do is find a cheap VOIP handset thats wireless |
| 06:52.07 | drmessano | HA |
| 06:52.13 | drmessano | SIP wireless = crap |
| 06:52.22 | drmessano | You're better off with an ATA and a 5.8GHZ phone |
| 06:52.49 | ipguy | drmessano: do you have real world experiance with wifi sip phones ? |
| 06:52.53 | tengulre | does the asterisk support PCM audio format? |
| 06:53.00 | drmessano | ipguy: do YOU? |
| 06:53.26 | ipguy | drmessano: mate, i'm just asking, don;t get defensive... |
| 06:53.47 | ipguy | drmessano: if you think my question are stupid, ignore me please |
| 06:53.51 | drmessano | done |
| 06:54.40 | ipguy | drmessano: thank christ fro that, it was like speaking to a 15yr old. |
| 06:55.17 | drmessano | Since youre a fucking smart ass expert-know-it-all after installing asterisk ONCE, glad to be of help |
| 06:55.55 | ipguy | lol, chill man, really... i'm simply asking questions here... :-) |
| 06:55.55 | drmessano | Seems like you're the one with the attitude, mate |
| 06:56.08 | drmessano | and being insulting, mate |
| 06:56.11 | ipguy | drmessano: me? really? howso ? |
| 06:56.27 | drmessano | "thank christ fro that, it was like speaking to a 15yr old." |
| 06:56.36 | drmessano | Anyway.. good luck |
| 06:57.31 | ipguy | luck, since when is luck required to setup anything these days ? |
| 06:57.32 | trnzmeta | err too much angst in here, chill guys, obama won remember |
| 06:58.02 | ipguy | i'm chilled, i think i just got drmessano at a bad time though |
| 06:58.13 | drmessano | Ah |
| 06:58.23 | drmessano | Of course.. insult someone and they are the bad guy |
| 06:58.27 | trnzmeta | it's the 3rd... it's about that time |
| 06:58.45 | drmessano | I won't be happy until obama nukes Australia.. whens inauguration? |
| 06:58.47 | ipguy | LOL.... reread your posts and then say that again |
| 06:59.06 | ipguy | drmessano: LOL, this is why everyone hates you yanks ! |
| 06:59.19 | drmessano | ipguy: I was just scrolling up to where I answered your questions and you called me a 15 yr old.. Did I misread, ass? |
| 06:59.43 | trnzmeta | pfft obama won't nuke australia him and our pm is like this |
| 06:59.47 | trnzmeta | *crosses fingers* |
| 06:59.49 | trnzmeta | mwhuauhauha |
| 06:59.54 | trnzmeta | this is getting interesting |
| 07:00.07 | drmessano | Not really |
| 07:00.24 | drmessano | Typical newb know-it-all.. doesnt get the answer he wants, so gets all crappy |
| 07:00.31 | drmessano | Blah blah blah |
| 07:00.33 | ipguy | ipguy>drmessano: do you have real world experiance with wifi sip phones ? |
| 07:00.33 | ipguy | [5:53pm] <drmessano> ipguy: do YOU? |
| 07:00.51 | drmessano | Indeed, I asked you a question |
| 07:01.13 | ipguy | ok, your a fool and i've waited to much time here... |
| 07:01.22 | trnzmeta | hahaha |
| 07:01.27 | drmessano | "you're" |
| 07:01.28 | trnzmeta | I can't believe you won |
| 07:01.35 | drmessano | I can't believe it either |
| 07:01.48 | joako | ipguy: can you recommend a quality WiFi SIP phone? |
| 07:01.57 | drmessano | I doubt it |
| 07:02.18 | drmessano | Since he was just asking about them himself |
| 07:02.24 | drmessano | and.. he's digging his bomb shelter |
| 07:02.31 | drmessano | I think trnzmeta made him wonder |
| 07:02.53 | joako | Or anyone for that matter can recommend a reliable one? |
| 07:02.59 | drmessano | Theres no such thing |
| 07:03.04 | trnzmeta | eh, don't bring me into this |
| 07:03.05 | drmessano | Get an ATA and a POTS cordless |
| 07:04.10 | joako | I was thinking SIP DECT phone |
| 07:04.14 | drmessano | trnzmeta: He was only trolling anyway.. Installed asterisk, now wants a PBX the size of a pack of playing cards for $25.. can't get it, so now he's gonna be an ass |
| 07:04.30 | drmessano | joako: Waste of time |
| 07:04.35 | joako | And I am legitimatly interested in a wireless solution |
| 07:04.42 | drmessano | joako: Get an ATA and a DECT phone.. forget the SIP part |
| 07:05.15 | joako | I have too many ATA'Ã unless you can tell me how to get port2 to work on HT-498 |
| 07:05.20 | jql | yeah, dect is a pretty decent way to go, considering the array of choices around |
| 07:05.55 | joako | i"ve connected a GXP-2000 to a WDS node and it worked as good as being directely wired, so voip over wifi seems possible |
| 07:05.59 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 07:06.01 | drmessano | Getting a SIP cordless is dumb.. you're tied to the sip implmentation in the phone.. and its pretty much deadware the second you buy it |
| 07:06.14 | drmessano | Get a well supported ATA and your choice of DECT phone |
| 07:06.27 | drmessano | VoIP over WIFI doesnt work |
| 07:06.43 | drmessano | Battery life, audio quality, etc |
| 07:07.03 | drmessano | ATA and a DECT phone is the best solution |
| 07:09.09 | drmessano | wonders why people bothering asking if they're just looking for someone to agree that their idea is correct |
| 07:09.14 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:13.17 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 07:13.27 | aiksa[LV] | morning everyone |
| 07:14.32 | hi365 | drmessano: are you just asking or are you looking for someone to agree that your idea is correct? |
| 07:14.44 | drmessano | douche` |
| 07:14.49 | drmessano | Wait |
| 07:14.55 | drmessano | touche` |
| 07:15.18 | hi365 | whats that, the new douche? |
| 07:16.27 | drmessano | Character from "In Living Color".. the one that was incarcerated that spoke very eloquently, but all his words were sexual references |
| 07:16.52 | drmessano | Used "douche`" instead of "touche`" |
| 07:16.59 | drmessano | Always stuck with me |
| 07:23.05 | drmessano | "oswald bates" |
| 07:23.16 | drmessano | That was the characters name |
| 07:23.20 | drmessano | Man I love YouTube |
| 07:26.23 | *** join/#asterisk marcrosoft (n=mark@75-175-248-134.hlna.qwest.net) |
| 07:27.04 | marcrosoft | Can I try out asterisk without hooking it up to PSTN or paying for a voip termination gateway? maybe just use softphone or something? |
| 07:27.28 | marcrosoft | I want to play with it before buying additional hardware or going all out |
| 07:28.20 | drmessano | yes you can |
| 07:29.05 | marcrosoft | so in a nutshell how would one set that up.. |
| 07:29.16 | marcrosoft | do you sign up to a freedialup connection and use a softphone? |
| 07:29.22 | drmessano | As you said, set it up with a couple softphones |
| 07:29.24 | marcrosoft | freeworlddialup |
| 07:29.27 | drmessano | No |
| 07:29.33 | drmessano | Use Asterisk |
| 07:29.35 | marcrosoft | k |
| 07:29.41 | marcrosoft | and how would i "dial" it |
| 07:29.52 | drmessano | You create extensions |
| 07:29.54 | drmessano | Its a PBX |
| 07:30.12 | drmessano | (to put it in simplest terms) |
| 07:30.29 | marcrosoft | oh |
| 07:30.36 | marcrosoft | so on the local network it would just be an extension |
| 07:30.39 | marcrosoft | no actuall number |
| 07:30.51 | drmessano | Right |
| 07:30.54 | marcrosoft | sweet |
| 07:31.06 | marcrosoft | and i could even call another computer on the network that had a softphone? |
| 07:31.16 | drmessano | Yep |
| 07:31.22 | marcrosoft | Alright thanks for your help |
| 07:31.27 | drmessano | No probs |
| 07:31.31 | marcrosoft | Im sure i'll be back :) |
| 07:31.32 | drmessano | ~book |
| 07:31.33 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 07:31.36 | drmessano | Grab that PDF |
| 07:32.04 | drmessano | It will tell you most of what you ever wanted to know or not know |
| 07:32.08 | *** join/#asterisk distatica (n=dist@unaffiliated/distatica) |
| 07:32.51 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 07:32.52 | marcrosoft | sweet a free oreilly book |
| 07:33.10 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 07:33.20 | marcrosoft | is voip over rated... |
| 07:33.27 | marcrosoft | i mean if you lose your internet connection you lose phones |
| 07:33.36 | tengulre | Does the asterisk support PCM audio format? |
| 07:33.42 | jql | definitely overrated, but not overpriced. :( |
| 07:33.56 | marcrosoft | true |
| 07:33.57 | [T]ank | getting an error... cant find where to kill the command... can anyone point me in the right direction? |
| 07:33.57 | [T]ank | ..................[Dec 3 07:33:25] WARNING[11296]: pbx.c:2981 ast_register_application: Already have an application 'Directory' |
| 07:34.01 | jql | tengulre: yes, in several ways |
| 07:34.05 | drmessano | First off, no one is making you use SIP |
| 07:34.05 | marcrosoft | practically free |
| 07:34.19 | marcrosoft | drmessano: i realize that you can hook it up to regular old pstn |
| 07:34.24 | marcrosoft | which I would probably do |
| 07:34.25 | drmessano | You can use SIP inside to connect to the PBX |
| 07:34.26 | jql | marcrosoft: very unfortunate, considering |
| 07:34.27 | marcrosoft | for reliability |
| 07:34.57 | marcrosoft | maybe you could have failover? |
| 07:35.12 | drmessano | or just get reliable internet :) |
| 07:35.17 | marcrosoft | hehe |
| 07:35.55 | marcrosoft | are the fxo fxs cards really that expensive? |
| 07:36.10 | drmessano | Depends on how many lines |
| 07:36.15 | drmessano | They can be |
| 07:36.21 | marcrosoft | 2-3 lines |
| 07:36.30 | drmessano | Couple hundred bucks |
| 07:36.39 | marcrosoft | cheaper than traditional pbx im sure |
| 07:36.54 | drmessano | yeah |
| 07:37.03 | marcrosoft | they look like a modem that is what makes them appear to be expensive |
| 07:37.11 | [T]ank | for the most part most carriers are sending calls to you via voip anyhow. its being converted to pstn at the "last mile" good internet and voip is just as reliable as pstn in my opinion. |
| 07:37.23 | distatica | My employer operates a small 40 room hotel, he requires a PBX with voice mail (including message indicator on handset), auto wake up calls (which must be a recorded message), auto attendant (for a missed call), CID, direct in dialing support, and the basic features. He currently has a Mytel system that is out of date with no support. |
| 07:37.39 | drmessano | or a "Mitel" |
| 07:37.44 | distatica | Would asterisk be a good place to be looking? |
| 07:37.50 | distatica | Oops, yes. |
| 07:37.55 | marcrosoft | [T]ank: good point |
| 07:38.00 | jql | an asterisk consultant... perhaps |
| 07:38.21 | marcrosoft | [T]ank: our internet service is the highest speed and it is through a telco company so... maybe it is just as good |
| 07:38.49 | marcrosoft | [T]ank: do you use QoS to ensure voice never chops? |
| 07:38.54 | drmessano | We use all VoIP and get 5 9's |
| 07:39.02 | drmessano | So not too bad |
| 07:39.17 | distatica | Are there any projects around asterisk that might focus more on this area? |
| 07:39.19 | *** join/#asterisk tshine (n=tshine@ip70-160-111-108.hr.hr.cox.net) |
| 07:39.41 | marcrosoft | how much bandwidth does a typically voip session take up? |
| 07:39.56 | drmessano | distatica: On what, making a PBX? That IS the project |
| 07:40.32 | drmessano | marcrosoft: Depends on codecs, etc |
| 07:41.01 | marcrosoft | rough guess what would you say you would need for average call quality |
| 07:41.09 | marcrosoft | for 1 session |
| 07:41.24 | jql | you need: zero packet loss, <10ms packet jitter |
| 07:41.34 | marcrosoft | ahh |
| 07:42.17 | marcrosoft | so interet to the gateway server has to be pretty solid |
| 07:42.17 | drmessano | It has to not be shit |
| 07:42.17 | distatica | drmessano: No, more focusing on hotel pbx systems, geared towards the hospitality industry. |
| 07:42.17 | jql | for quality, yes |
| 07:42.26 | distatica | ie. wake up calls are not a common feature requested on office pbx systems. |
| 07:42.27 | jql | distatica: hospitality voip asterisk has some hits on google |
| 07:42.46 | jql | the hospitality systems require special billing, which makes them somewhat specialized |
| 07:43.04 | drmessano | Hire a consultant |
| 07:43.21 | distatica | Ramada, well isn't that interesting. heh |
| 07:43.22 | jql | I've looked into what it would take to do hotel systems, and I discovered that I hate hotels |
| 07:43.25 | drmessano | The more you ask, the more it's obvious you're over your head |
| 07:44.49 | distatica | What can I expect to pay for a consulatant on something like this? Roughly. |
| 07:44.54 | distatica | consultant yet. |
| 07:45.32 | jql | only knows what he would charge... |
| 07:45.45 | distatica | I have to make sure it's more cost effective to go that route, otherwise we'll just order the installed, ready to go system. Seems to me Asterisk is making an impact somewhere though, either it's just that damn good, or it's cheaper in the end. |
| 07:45.56 | drmessano | $1000-$10000, give or take $10000-$20000 |
| 07:46.48 | distatica | for something like that, what would I be looking at in hardware? |
| 07:46.59 | drmessano | .... |
| 07:46.59 | jql | how many handsets? |
| 07:47.08 | jql | that's the real cost sink, there |
| 07:47.19 | distatica | 40 rooms, 1 office, and two would be cordless. |
| 07:47.24 | distatica | yeah, I wondered about that. |
| 07:47.58 | distatica | I considered voip, but that's rough when you consider what is required in the reciever. (unless people use their laptops, and no, we're not allowed to do that. ) |
| 07:48.13 | jql | so, at least 50 handsets @ ~200 bucks each, depending on your needs |
| 07:48.14 | distatica | something like this seems like it would fit the bill: http://www.engadget.com/2007/03/19/fonality-launches-trixbox-appliance-asterisk-based-voip-pbx/ |
| 07:48.23 | drmessano | ROFL |
| 07:48.30 | drmessano | Did someone say trixbox? |
| 07:48.41 | distatica | POS? |
| 07:48.53 | drmessano | Trixbox is the windows vista of VoIP |
| 07:48.56 | drmessano | Run, run far |
| 07:49.16 | distatica | I see |
| 07:49.25 | marcrosoft | and why would we do that when we have our persoinal helper drmessano :) |
| 07:49.44 | jql | the actual pbx server itself will be a trivial expense compared to the rest |
| 07:50.05 | jql | which is actually why the PBX vendors feel so justified raping you for their hardware |
| 07:50.07 | marcrosoft | jql: mostly in the handsets? |
| 07:50.10 | drmessano | distatica: Unless you plan to sink a LOT of money into uneducated moves, you need to hire this out.. you're window shopping |
| 07:50.22 | distatica | In the end, after my handsets, consultant fees, installation, training, all that, IF you were to actually contract out like that, what would we be looking at? |
| 07:50.35 | distatica | drmessano: that's precisely what I'm asking about. |
| 07:50.49 | distatica | BUT, it would be ridiculous to bring someone in at that price, only to go with a mitel. |
| 07:50.53 | jql | from my experience, expenses run like handsets > wiring > internet > pbx hardware for large values of handsets |
| 07:50.53 | distatica | that's called dumb. |
| 07:51.16 | drmessano | You're asking for someone that cant be quoted in an IRC channel |
| 07:51.22 | drmessano | something* |
| 07:52.11 | distatica | you're telling me, there is not one person in here that has done a similar type contract, or is a contractor, that would know what their rates are; and be able to tell me, for the purposes of getting a rough estimate. |
| 07:52.12 | drmessano | You want to give blueprints for cable runs, photos of the wiring closet, a spreadsheet with specific feature needs, employee training requirements, then talk about handset needs, etc |
| 07:52.17 | marcrosoft | jql do you usually use the same network for your phones as your computers? or have seperate cat5? |
| 07:52.34 | drmessano | Depends on the size |
| 07:52.39 | distatica | true |
| 07:52.45 | jql | generally different, powered via a PoE switch |
| 07:52.51 | drmessano | Small network, phones are fine on the network |
| 07:52.55 | jql | when practical |
| 07:53.20 | drmessano | Above 15 phones, you need to consider segmenting (roughly) |
| 07:53.31 | marcrosoft | i see |
| 07:53.42 | drmessano | Assuming 15 users of course |
| 07:53.44 | marcrosoft | the university i used to work with had seperate network.. so i was wondering |
| 07:53.48 | marcrosoft | they had hundreds of phones |
| 07:53.52 | marcrosoft | obviously |
| 07:54.04 | drmessano | Would most definitely be on a seperate network |
| 07:54.18 | jql | oh, in a university it's a must. college kids will rape your network |
| 07:54.23 | marcrosoft | lol |
| 07:54.24 | marcrosoft | well |
| 07:54.27 | jql | case in point: bittorrent |
| 07:54.32 | marcrosoft | dorms were in an entirely different segment |
| 07:54.39 | marcrosoft | than departments |
| 07:54.43 | jql | ahh |
| 07:54.43 | drmessano | Even still |
| 07:54.54 | marcrosoft | and they had paketeer on the edge |
| 07:54.55 | drmessano | The phones would dictate the necessity |
| 07:54.58 | marcrosoft | shaping all traffic |
| 07:55.02 | marcrosoft | ofcourse |
| 07:55.21 | marcrosoft | well i need to go to bed.. thanks for helping out |
| 07:55.31 | marcrosoft | jql and drmessano |
| 07:55.32 | jql | I just find it easier to deal with a separate network. less QoS worries, easier to battery backup, etc |
| 07:55.37 | drmessano | yeah |
| 07:55.58 | drmessano | Youre not over supporting the data network to keep voice up |
| 07:57.39 | drmessano | Im in the process of converting an EMA emcomms vehicle from cellular to VoIP |
| 07:57.56 | drmessano | Easier dealing with reliable internet than cell service |
| 07:57.59 | drmessano | and more flexible |
| 07:59.56 | drmessano | Sad and ironic part is that cell service has desensitized us to phone problems |
| 08:00.14 | drmessano | so 99.95% uptime isn't so bad |
| 08:00.31 | drmessano | Which in reality is kinda crappy |
| 08:00.53 | drmessano | But you consider cell service, and thats like high availablity |
| 08:05.46 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 08:08.46 | orkid | same with ipod/mp3 |
| 08:08.57 | orkid | but that kidna shizzo only cuts it with some |
| 08:09.12 | orkid | and u pay with quality for convenience |
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| 08:32.49 | fcois93 | hello all ! |
| 08:33.26 | fcois93 | I have a problem in my asteriskS servers |
| 08:33.44 | fcois93 | can you see at my topic here http://www.asterisk-france.net/community/showthread.php?t=6696 ? |
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| 08:38.47 | fcois93 | my problem is here:http://forums.digium.com/viewtopic.php?p=121341#121341 |
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| 08:50.08 | tengulre | which g729 codecs is good for asterisk? |
| 08:50.57 | mort_gib | tengulre: Digium |
| 08:51.13 | jql | the USian-legal one? |
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| 09:02.59 | hi365 | anyone auto provisioning snom's? |
| 09:06.34 | fcois93 | I have a probleme with my new network http://forums.digium.com/viewtopic.php?p=121341#121341 |
| 09:06.37 | fcois93 | please help me |
| 09:09.53 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:12.07 | aiksa[LV] | i have a question - how could I catch transfer event in ami? is there a special event for that or another Link event would appear for the transfered call? |
| 09:12.23 | aiksa[LV] | asterisk - 1.4 |
| 09:17.02 | aiksa[LV] | is it true that AMI was introduced Transfer event only in 1.6 ? |
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| 09:30.57 | mort_gib | hi365: Still same problems? |
| 09:31.13 | *** join/#asterisk donnib (n=aaaa@0x555281d0.adsl.cybercity.dk) |
| 09:31.57 | donnib | hi |
| 09:32.07 | hi365 | mort_gib: same as? I just cant seem to hack the provisioning files... I belive I got the "general" file (snom320.htm) but I cant seem to figure you format for the individueal files (snom320-MAC.html) here is the pastebin: |
| 09:32.29 | donnib | i have asked this question before and i discussed this with some of you and i am asking it again, maybe new ideas come up :) |
| 09:32.52 | donnib | i have problem with a SPA942 not be able to register some times, i have other clients using Xlite and not having one single problem |
| 09:33.08 | *** join/#asterisk invalidrecord (n=fares@87.112.10.227.plusnet.ptn-ag1.dyn.plus.net) |
| 09:33.25 | donnib | i just reg failed on thr SPA942 |
| 09:33.38 | hi365 | mort_gib: http://pastebin.ca/1274772 |
| 09:33.46 | invalidrecord | hi im having a prodlem getting res_pgsql to be built, if i compile in ubuntu it gets built but on my mac it dosent |
| 09:33.53 | donnib | i leaning towards a hardware problem but can't really know since i don't have another phone to try with |
| 09:34.11 | invalidrecord | i am passing --with-postgres=/opt/local/lib |
| 09:38.32 | fcois93 | I have that error "chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to" |
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| 10:11.29 | Faustov | hi, is there any document describing the migration from zaptel to dahdi? (no hw, just soft, i use it only for meetme) |
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| 10:25.17 | Faustov | hmm seems not, just load chan_dahdi.so instead of chan_zap.so which is no longer built, even if --with-zaptel is set and --with-dahdi is not set during configure |
| 10:25.33 | Faustov | might be a good idea to add this to the wiki :) |
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| 10:31.46 | angryuser | when i receive a call i am setting a ${CALLERID(num)} to a $var i want in my agi php script, but still the original number is displayed on my desk phone, i would like it to change to $var here is the script http://www.pastebin.ca/1274799 line 50, the question is, maybe i am missing something in SET with agi? , help ;) |
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| 10:34.25 | angryuser | here is the asterisk output http://www.pastebin.ca/1274803 i need to see '32' in this case, but still i have 08xxxxxxxxxx |
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| 10:45.31 | key2 | is 1.6.0.1 a release or a beta ? |
| 10:49.47 | tzafrir_laptop | A release |
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| 11:00.08 | angryuser | ok it is working now ;) |
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| 11:23.36 | xacatecas | hi all, i note that the new asterisk-addons (1.6.0) changed the way in which the column names for the sql query is constructed. It now seems to query what columns are in the table and tries to insert something into all of them. |
| 11:23.44 | xacatecas | is there any way to tell it to leave some columns alone? |
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| 12:46.43 | yang | I am wondering about the thing called session border controller and does it have a differnt name in linux ? |
| 12:47.07 | yang | Someone suggested it as a good thing for VOIP |
| 12:49.05 | coppice | its a good thing for people selling session border controllers |
| 12:51.54 | yang | those tend to be qutie expenssive |
| 12:52.40 | coppice | OK, its a *really* good thing for people selling session border controllers |
| 12:57.13 | espent | hi |
| 12:57.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:57.41 | espent | how do i pass variables to asterisk through $agi->exec('Dial', 'number' |
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| 13:23.29 | dominic1 | after installing dahdi and recompiling asterisk, how can I remove zaptel? |
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| 13:37.09 | angryuser | dominic1: your dahdi channels are not working ? |
| 13:37.26 | dominic1 | my channels are working |
| 13:37.31 | dominic1 | anything is okay |
| 13:37.40 | dominic1 | I just want to remove the zaptel stuff |
| 13:38.48 | chazz | the zaptel tools or the modules? |
| 13:39.05 | dominic1 | both |
| 13:39.27 | saftsack | there is a tool what logs what make install does |
| 13:39.45 | saftsack | with this tool you can remove the installed data. that means you have to reinstall it and then you can remove it |
| 13:40.59 | saftsack | e.g. you can try debians m_pkg |
| 13:41.18 | saftsack | create a m_pkg, install it and remove it |
| 13:41.30 | dominic1 | what's the name of the tool? make uninstall? |
| 13:41.55 | saftsack | http://www.linuxfromscratch.org/hints/downloads/files/PREVIOUS_FORMAT/install-log.txt |
| 13:42.08 | saftsack | this is a nice tool. read this manual and you can go |
| 13:42.43 | dominic1 | thank you very much. But there is no integrated mechanism in zaptel to uninstall? |
| 13:42.54 | [TK]D-Fender | dominic1: No |
| 13:43.02 | dominic1 | :-( |
| 13:43.05 | saftsack | no the makefile doesnt include an uninstall routine |
| 13:43.37 | saftsack | but with the lfs make install logfile creator you can do the same job in maybe 2 minutes |
| 13:44.03 | dominic1 | okay, thank you very much |
| 13:46.26 | xacatecas | hi all, i note that the new asterisk-addons (1.6.0) changed the way in which the column names for the sql query is constructed. It now seems to query what columns are in the table and tries to insert something into all of them. |
| 13:46.27 | xacatecas | is there any way to tell it to leave some columns alone? |
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| 14:25.54 | maxhbp2005 | Hi |
| 14:25.54 | *** join/#asterisk DarkRift (n=dark@65.92.166.196) |
| 14:26.30 | maxhbp2005 | i want to prevent 302 temporarily moved message from asterisk |
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| 14:26.51 | maxhbp2005 | is there any configuration in asterisk not to do forward to another user |
| 14:26.58 | maxhbp2005 | ? |
| 14:27.07 | maxhbp2005 | please help me regarding this |
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| 14:30.01 | [TK]D-Fender | maxhbp2005: AFAIK there is no way to prevent a SIP devices from being allowed to transfer a call |
| 14:30.21 | maxhbp2005 | ok thanks for your reply |
| 14:30.31 | maxhbp2005 | means we have to set that in my linksys device |
| 14:30.36 | maxhbp2005 | which i am using |
| 14:32.40 | maxhbp2005 | [TK]D-Fender: please correct me if i am wrong |
| 14:33.15 | [TK]D-Fender | maxhbp2005: The answer is clear. If * can't stop you, your device had better be able to restrict itself |
| 14:33.29 | [TK]D-Fender | maxhbp2005: The kind of question that really never needs asking |
| 14:34.26 | Katty | [TK]D-Fender: fender, can you connect to me on port 34? |
| 14:34.50 | maxhbp2005 | ok fine |
| 14:34.58 | [TK]D-Fender | Katty: Whats that? |
| 14:35.06 | Katty | [TK]D-Fender: register a sip phone |
| 14:35.11 | Katty | [TK]D-Fender: on port 34, rather than 5060 |
| 14:35.23 | [TK]D-Fender | Katty: can't really concentrate on stuff like that at the office... |
| 14:35.38 | [TK]D-Fender | Katty: You need to get yourself a remote server for testing :) |
| 14:35.40 | Katty | k, i'll get someone else (= |
| 14:36.02 | [TK]D-Fender | feels used and dirty... |
| 14:36.09 | Katty | hugs [TK]D-Fender |
| 14:36.13 | [TK]D-Fender | showers and cries |
| 14:36.13 | anonymouz666 | [TK]D-Fender: if (is_method("REFER")) { drop; } |
| 14:36.16 | anonymouz666 | ;) |
| 14:36.53 | [TK]D-Fender | anonymouz666: Yes, source code is inherently modifyable.... just looking at it from a user/admin vs coder |
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| 14:42.47 | disposable | i have 2 queues with roundrobin strategy. however, when a new call comes in, it doesn't start with ringing the first phone in the queue, but instead the one after the last one that picked up a call. my queues.conf is here http://pastebin.com/d5a9a912e Can somebody please tell me how to change it to do what i want? |
| 14:43.22 | disposable | it's asterisk 1.2 |
| 14:44.08 | tokozedg | is there any web client for asterisk like X-lite but web workink on sip? |
| 14:44.30 | [TK]D-Fender | disposable: Absolutely sure its 1.2? |
| 14:45.48 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:46.00 | disposable | [TK]D-Fender: yes |
| 14:47.12 | [TK]D-Fender | disposable: Ok, because in 1.4 they resync'd it to be synonymous with rrmemory... |
| 14:47.38 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 14:47.42 | disposable | it's 1.2.7.1-bristuff |
| 14:48.34 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 14:49.13 | [TK]D-Fender | disposable: pastebin a channel dump followed by a call going into the queue (verify that the users are available) |
| 14:49.19 | *** join/#asterisk synchris (n=synchris@athedsl-4381731.home.otenet.gr) |
| 14:50.22 | *** join/#asterisk mog (n=mog@nat/digium/x-c34926aba75464a6) |
| 14:50.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:50.47 | disposable | [TK]D-Fender: it's a customer so this may take a few minutes |
| 14:51.16 | *** join/#asterisk oktay (n=oktay@81.215.202.193) |
| 14:51.40 | oktay | hi. i forget how to call voicemail.. :( |
| 14:52.10 | jaytee | VoicemailMain |
| 14:52.47 | [TK]D-Fender | oktay: "exten => 1234,1,Voicemail(aboxgoeshere) |
| 14:53.15 | [TK]D-Fender | jaytee: tsk, tsk... |
| 14:53.25 | tokozedg | or exten => 1234,1,VoicemailMain(${CALLERID(num)}) |
| 14:53.34 | jaytee | yeah, I was thinking he meant call into voicemail |
| 14:53.41 | jaytee | not leave voicemail |
| 14:53.42 | [TK]D-Fender | tokozedg: And now ASSUMING the CID is of value.. |
| 14:53.45 | Katty | yay jaytee will help me! |
| 14:53.48 | [TK]D-Fender | The trench deepens! |
| 14:53.55 | oktay | thanks guys |
| 14:54.13 | Katty | jaytee: i has a QUEST for you |
| 14:54.14 | *** join/#asterisk ChkDigit (n=mike@24.72.71.175) |
| 14:54.56 | jaytee | so does anyone have any DID numbers in Tokyo I can route all the outgoing calls from our HR department to? I'm cutting them over to Asterisk today. |
| 14:55.01 | jaytee | :-) |
| 14:58.12 | *** join/#asterisk mv2 (n=mv2@83.240.229.38) |
| 14:58.22 | mv2 | any help with cisco ip phones ? |
| 14:58.36 | tzanger | http://pages.ripco.net/~havoc/elven-love-slave.html |
| 14:58.56 | tzanger | that is at least 12 years old |
| 14:59.38 | tzanger | citats's brother sent that to me YEARS ago |
| 15:00.08 | mv2 | any help with cisco ip phones ? |
| 15:01.45 | *** join/#asterisk maziah (i=maziah@203-97-204-234.dsl.clear.net.nz) |
| 15:02.09 | [TK]D-Fender | mv2: maybe if you asked a SPECIFIC question. |
| 15:02.10 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 15:02.42 | mv2 | Fender: I to restore the default configs on a ip phone 7961 |
| 15:03.13 | mv2 | I tried to upgrade the phone but now the phone doesn't work |
| 15:03.47 | jaytee | "Would you like your firmware deep fried? Or broiled?" |
| 15:05.34 | mv2 | Fender:any help? |
| 15:06.01 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:06.45 | kb3ien | Microwaved... |
| 15:08.44 | [TK]D-Fender | mv2: "doesn't work" is a not a description anyone can help you with. |
| 15:08.53 | *** join/#asterisk cjk (n=cjk@vodsl-9252.vo.lu) |
| 15:09.42 | cjk | hi, if i have an extension _89X. in the users context and 898910 in an included context and my user calls 898910. which extension will be matched? |
| 15:09.55 | mv2 | doesn't show up anything on LCD |
| 15:10.21 | mv2 | the phone tries to get some files from ftpd |
| 15:10.28 | [TK]D-Fender | mv2: You realize you're not saying ANYTHING of value here, right? |
| 15:10.34 | *** join/#asterisk packetstream (n=chatzill@77.240.56.22) |
| 15:10.45 | packetstream | Hi All |
| 15:10.52 | mv2 | Fender: ? |
| 15:10.58 | [TK]D-Fender | mv2: Go try & flash your firmware again. |
| 15:11.19 | mv2 | hey fender you are a smart guy |
| 15:11.24 | mv2 | thanks |
| 15:11.33 | mv2 | If i could find the firmware |
| 15:11.38 | [TK]D-Fender | mvAnd make sure you are using a proper compatible firmware & set of configs |
| 15:12.10 | mv2 | thats not WHAT I'M ASKING FOR |
| 15:12.18 | mv2 | Dont have the firmware |
| 15:12.25 | [TK]D-Fender | mv2: www.cisco.com <- |
| 15:12.29 | mv2 | and want to get things back |
| 15:12.48 | [TK]D-Fender | mv2: and if you've flushed your phones configs, there is no getting your settings back |
| 15:13.13 | packetstream | Can anyone recommend a resources for dealing with Grandsteam issues. The grandsteam site is no help at all and neither is google at the moment. Perhaps it's the search terms I've used. Anyway I need to find out why 3 phones all of a sudden do the following: |
| 15:13.34 | packetstream | the phone makes a loud cracking sounds, phone itself goes off - all lights shine, it dies and a few minutes later it comes to life and reboots |
| 15:13.36 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-e2b1eec7b87aaaea) |
| 15:13.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:14.07 | packetstream | any search terms for google would be appreciated? |
| 15:15.08 | packetstream | tried grandsteam reboots/all lights flash/buzzing/etc |
| 15:15.45 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 15:16.13 | *** join/#asterisk arpu (n=arpu@chello080109017021.12.14.vie.surfer.at) |
| 15:17.22 | packetstream | I though it might be a faulty power supply but 3 out of 8 phones doesn't seem likely |
| 15:18.11 | kb3ien | can one spec sounds that arn't in '/var/lib/asterisk/sounds/' to be played by Background() ? |
| 15:18.42 | jaytee | ~grandstream |
| 15:18.43 | jbot | it has been said that grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 15:19.37 | packetstream | lol |
| 15:19.39 | lmadsen | funny how I just saw a show on Yugo's that still exist and have been suped up in Yugoslavia |
| 15:19.56 | lmadsen | kb3ien: just provide the path to the sound file |
| 15:20.05 | lmadsen | Background(/path/to/file) |
| 15:20.19 | kb3ien | it abides by the /, cools beans. |
| 15:20.48 | jaytee | I have 4 Grandstream phones and 32 Polycom phones. Of all those phones the ONLY problems I've had are with Grandstreams. I keep 2 spare AC/DC power supplys handy |
| 15:21.19 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 15:21.46 | *** join/#asterisk RobertLaptop (n=rmiddle@pool-70-22-18-194.balt.east.verizon.net) |
| 15:22.46 | packetstream | Jaytee I hate Polycom phones |
| 15:22.53 | *** join/#asterisk pecanha (n=e@189.106.175.43) |
| 15:23.39 | packetstream | they need to be rebooted again and again just to install |
| 15:23.44 | jaytee | packetstream, then you must really love echo and jitter :-) |
| 15:23.51 | packetstream | lol |
| 15:24.53 | jaytee | I had to upgrade the firmware on the Grungestreams just to get silence suppression to turn the frak off. It would say it was off but it wasn't. To get continuous MOH while on hold I had to blow into the handset mic constantly. |
| 15:25.29 | packetstream | lol |
| 15:25.41 | jaytee | and I found that info BURIED in a firmware update text file where you'd be not likely to come across it. |
| 15:26.27 | jaytee | packetstream, but go right ahead and support the commie bastards with their slave labor junk and by the way, I hear WalMart's having a major sale this week too! |
| 15:27.09 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:27.13 | coppice | as opposed to use the made in china polycoms? |
| 15:27.13 | lmadsen | today is 'hug a commie' day |
| 15:27.25 | packetstream | lol |
| 15:28.33 | *** join/#asterisk dkatz333 (n=dkatz@mail.stonehilltaylor.com) |
| 15:28.56 | dkatz333 | Hello all |
| 15:28.58 | jaytee | polycom's are made in china? Oh CRAP!!!! We're doomed!!! commies are takin over the damn world. |
| 15:29.21 | dkatz333 | I have an interesting issue that I think someone may have solved already. |
| 15:29.23 | jaytee | How can a phone of such good quality come from China? It doesn't make sense. |
| 15:30.01 | lmadsen | "http://www.moviesoundscentral.com/wavs/armageddon/armageddon8.wav |
| 15:30.05 | coppice | in previous recessions production has moved rapidly to china. this time I think things will be different. there's very little production left to move to china |
| 15:30.25 | jaytee | says on the box for one of the new 330's I'm rolling out that it was made in Thailand. Pffffffft! so there. whew! |
| 15:30.32 | packetstream | åè¯/è¯èª, |
| 15:30.37 | lmadsen | Lev: "American components, Russian Components, ALL MADE IN TAIWAN!" -- quote from Armageddon |
| 15:30.44 | dkatz333 | Using asterisk, i have about 50 Grandstream 2020's all using early dial (484). If I setup a "hint" for thier SIP account, it generates enormous traffic when the user is dialing, as a result of the "invite" and then 484 messages. Anyone have a solution? |
| 15:31.26 | lmadsen | oops.. maybe I should have checked if the link worked before posting :) |
| 15:31.34 | dkatz333 | I've used DevState to get it working for now, but it's not smooth and sometimes the user is on a call and the DevState shows available. |
| 15:31.45 | coppice | packetstream: who says åè¯/è¯èª ? that's very odd chinese |
| 15:32.16 | packetstream | I learn chinese just in case |
| 15:32.27 | packetstream | learn=learning |
| 15:32.42 | packetstream | I=I'm |
| 15:32.51 | Katty | weeeeeeeeeeee |
| 15:33.22 | coppice | packetstream: ä½ çä¸æä¸å¥½ :-) |
| 15:33.23 | packetstream | æ£é³æ¸é¢; S:æ£é³ä¹¦é¢ |
| 15:33.50 | *** join/#asterisk clintc (n=clintc@n128-227-5-240.xlate.ufl.edu) |
| 15:34.55 | packetstream | Ok I'm cutting and pasting chinese text from a wiki :-) |
| 15:35.23 | coppice | but è¯èª is still an odd expression |
| 15:35.45 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 15:36.08 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 15:36.22 | pecanha | guys... I can't receive calls -> my extension. I can put in hold, I can call between extensions. And even tried from network with and without NAT (changing according nat=yes and nat=no). SIP peer is registered... I already tried everything I know. :/ |
| 15:36.44 | asteriskmonkey | any one seen this error before ? Dec 3 05:24:27 WARNING[4312]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0xb7809c40 (len 469) to 192.168.16.62:5060 returned -1: Bad file descriptor |
| 15:37.45 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-64c0ca5622bf1b97) |
| 15:37.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:37.51 | packetstream | coppice: I got the text from this line: Chinese or the Sinitic language(s) (æ±è¯/æ¼¢èª, pinyin: Hà nyÇ; åè¯/è¯èª, HuáyÇ; or ä¸æ, ZhÅngwén) |
| 15:38.57 | coppice | æ±è¯/æ¼¢èª == normal, ä¸æm == normal, åè¯/è¯èª == odd |
| 15:46.42 | asteriskmonkey | noone seen this before ? Dec 3 05:31:22 WARNING[4312]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x906f098 (len 469) to 192.168.16.62:5060 returned -1: Bad file descriptor |
| 15:46.50 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:47.41 | dkatz333 | not I, sorry monkey |
| 15:49.54 | *** join/#asterisk DeVilSoulBlacK (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 15:50.00 | packetstream | coppice are you saying the wiki is using the wrong symbols or just that the text is grammatical wrong |
| 15:50.02 | packetstream | ? |
| 15:50.37 | coppice | its obvious what it means, but I've never heard anyone use that expression |
| 15:51.39 | DeVilSoulBlacK | hi, ist possible disable the user dial via softphone the numbers ?, |
| 15:56.21 | disposable | [TK]D-Fender: having done some testing and reading up on the issue, i have discovered that the fact RoundRobin call queue strategy doesn't reset for each call is a feature and not a bug. thanks for the help anyway. |
| 15:56.52 | packetstream | ok what does the expression mean ? |
| 15:58.07 | *** join/#asterisk asterisk` (i=xpl@84.126.215.171.dyn.user.ono.com) |
| 15:58.52 | asterisk` | hi there |
| 15:59.33 | [TK]D-Fender | disposable: Should be a bug. rrmemory remembers where you leave off, roundrobin does not. |
| 16:00.50 | asterisk` | [TK]D-Fender : do you have a second ? |
| 16:02.06 | *** join/#asterisk Chesther (n=cam2@cam2-win.cit.cornell.edu) |
| 16:05.40 | [TK]D-Fender | asterisk`: ask in public |
| 16:06.51 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 16:09.13 | *** join/#asterisk stevie[xxx] (n=stevie@e177156199.adsl.alicedsl.de) |
| 16:10.12 | *** part/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 16:10.52 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 16:12.40 | DeVilSoulBlacK | hi, ist possible disable the user dial via softphone the numbers ?, (via dialplan or context ) |
| 16:14.02 | *** join/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:15.04 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) [NETSPLIT VICTIM] |
| 16:15.04 | farah | #join cacti |
| 16:15.04 | farah | oops |
| 16:15.09 | *** join/#asterisk zeljkoMON (n=bum@cable-89-216-173-176.dynamic.sbb.rs) |
| 16:15.23 | lmadsen | atleast you weren't trying to join some fetish room :) |
| 16:15.32 | zeljkoMON | hi all |
| 16:15.47 | farah | lol:) |
| 16:15.48 | Faustov | cacti is fetish |
| 16:16.01 | farah | anyone can help me with cacti plz? |
| 16:16.12 | farah | i want to monitor snmp with cacti |
| 16:16.23 | *** join/#asterisk Corydon76-dig (i=two@pdpc/supporter/bronze/Corydon76-home) |
| 16:16.23 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) [NETSPLIT VICTIM] |
| 16:16.23 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 16:16.23 | *** join/#asterisk RobH_DC (n=RobH@rob.tech.wikimedia.org) |
| 16:16.23 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) [NETSPLIT VICTIM] |
| 16:16.23 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) [NETSPLIT VICTIM] |
| 16:16.23 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 16:16.23 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 16:16.24 | *** join/#asterisk [netman] (n=netman@181.Red-88-17-242.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk LeddyHM (n=none@polar.artica.net) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 16:16.24 | *** join/#asterisk miloux (i=milu@213.88.194.123) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk sh0tt (n=sh0t@83.19.145.67) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) [NETSPLIT VICTIM] |
| 16:16.24 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 16:16.24 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) [NETSPLIT VICTIM] |
| 16:16.24 | *** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net |
| 16:16.24 | coppice | yeah, but its not really perverted like, say, join #sarahpalin |
| 16:16.25 | zeljkoMON | can some1 help me with including context? |
| 16:16.34 | [TK]D-Fender | farah: WRONG CHANNEL |
| 16:16.58 | farah | where do i have to ask my question? |
| 16:17.08 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 16:18.23 | [TK]D-Fender | farah: ... this isn't #cacti |
| 16:18.38 | zeljkoMON | does contexts inherit includes form included context? |
| 16:18.44 | lmadsen | yes |
| 16:18.45 | farah | there is no cacti this channel doesnt exist |
| 16:18.47 | [TK]D-Fender | zeljkoMON: Yes |
| 16:19.01 | zeljkoMON | is there any way i can make set up like this |
| 16:19.08 | zeljkoMON | have 3 contexts |
| 16:19.11 | zeljkoMON | 1 local |
| 16:19.15 | [TK]D-Fender | farah: I'm fairlycetain there is no #fordmotorco but that doesn't mean we support your CAR here either... |
| 16:19.23 | zeljkoMON | 1 internet and 1 for outbound calls |
| 16:19.40 | zeljkoMON | and enble only local users to use outgoing calls |
| 16:20.07 | [TK]D-Fender | zeljkoMON: then make sure only your phone's include contexts with outbound access |
| 16:20.10 | farah | [TK]D-Fender: it's related to asterisk snmp!but anyway..thank you |
| 16:20.50 | zeljkoMON | yea but the problem is that both local and net users can talk to eachother |
| 16:21.52 | [TK]D-Fender | zeljkoMON: pastebin is your friend... show us your dialplan. |
| 16:21.57 | [TK]D-Fender | !~pb |
| 16:21.59 | [TK]D-Fender | ~pb |
| 16:21.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:22.01 | [TK]D-Fender | ^^^^^^^^^^ |
| 16:22.25 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 16:23.23 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:23.26 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:23.57 | SQLDarkly | Anyone know where there is some documentation even minimal on replacing the mini http server with an Apache server different maching.. |
| 16:25.49 | SQLDarkly | machine that was ;) |
| 16:26.41 | zeljkoMON | [TK]D-Fender prob is i aint o that machine |
| 16:27.00 | zeljkoMON | just need a way to include all like it should be |
| 16:27.40 | zeljkoMON | can i make new context for excample tat will have net and local included so they can talk to eachother |
| 16:27.49 | zeljkoMON | and one with local and outbound? |
| 16:27.58 | [TK]D-Fender | zeljkoMON: Of course |
| 16:28.54 | DeVilSoulBlacK | hi, ist possible disable the user dial via softphone the numbers ?, (via dialplan or context ) |
| 16:29.20 | *** part/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:30.27 | *** join/#asterisk Ast001 (n=uros@cable-94-189-185-5.dynamic.sbb.rs) |
| 16:30.49 | [TK]D-Fender | DeVilSoulBlacK: "the numbers"? WHT "numbers"? |
| 16:32.07 | DeVilSoulBlacK | the number was in the softphone |
| 16:32.23 | DeVilSoulBlacK | 0 1 2 3 4 5 6 7 8 9 |
| 16:32.41 | [TK]D-Fender | DeVilSoulBlacK: Still makes no sense. What are they doing if they aren't allowed to dial? |
| 16:33.35 | DeVilSoulBlacK | because i have i predict call via php |
| 16:34.08 | DeVilSoulBlacK | and i dont need the agent can dial via softphone |
| 16:34.40 | DeVilSoulBlacK | because the predict script make the calls(dials) |
| 16:34.54 | Ast001 | Hi, I've got following error on Asterisk CLI and Digium TE121 card : WARNING pri err on span 0 we think we are cpe but they think they are too |
| 16:35.35 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:35.39 | Ast001 | ERROR chan_zap.c ZT_PRI_ERROR!! got ! frame while link state 8 |
| 16:36.14 | Ast001 | I made loopback RJ-45 and put in card and card is green but that annoying message show up insted of restart channel 1... 30 |
| 16:36.48 | coppice | its not an annoying message. its a clear and important one. you are just failing to read it |
| 16:37.02 | [TK]D-Fender | DeVilSoulBlacK: then point them to a context with no usable extens in it |
| 16:37.39 | *** join/#asterisk zeljkoMON (n=bum@cable-89-216-173-176.dynamic.sbb.rs) |
| 16:38.58 | Ast001 | I know it is important one but I am not sure how clear it is |
| 16:39.27 | Ast001 | is it zaptel driver problem ? libpri problem or what ? |
| 16:39.37 | coppice | How much clearer than "we think we are cpe but they think they are too" do you expect to get? |
| 16:40.00 | zeljkoMON | it doesnt seem to work |
| 16:40.06 | zeljkoMON | here is my conf http://pastebin.com/d7b8c6cd5 |
| 16:40.10 | Ast001 | who are they ? There are only me card and rj45 |
| 16:40.40 | coppice | you said you looped two ports together. you've configured them both as CPE |
| 16:40.46 | Ast001 | pri device is not connected at all |
| 16:41.47 | Ast001 | no I have only 1 port on card |
| 16:41.59 | Ast001 | and I made loopback for it |
| 16:42.14 | coppice | so you looped one port back to itself, and its receiving its own output. |
| 16:43.04 | Ast001 | it should reset channels not showing error message right ? |
| 16:44.06 | Ast001 | something like reconfigure or reset channel 1 .. 30 |
| 16:44.45 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:45.04 | [TK]D-Fender | zeljkoMON: show us this user placing a call they shouldn't be able to (CLI output), and thier sip.conf config (or whatever they use) |
| 16:45.44 | mikealeonetti | is there something like announcements but for just music on hold? just something to let the customer know he isn't just listening to music but is on hold? |
| 16:45.55 | zeljkoMON | [TK]D-Fender in this config |
| 16:46.07 | zeljkoMON | local and net should be able to communicate? |
| 16:46.20 | Ast001 | is libtermcap-devel package essential for building libpri or zaptel ? There is no such package on Ubuntu |
| 16:46.45 | [TK]D-Fender | zeljkoMON: CONTEXTS don't "communicate" |
| 16:47.23 | [TK]D-Fender | zeljkoMON: And those 2 contexts do not include any others |
| 16:48.48 | Ast001 | Is compiling order 1) libpri then 2)zaptel then 3)asterisk ok or I need to compile zaptel first libpri second and asterisk third ? |
| 16:49.32 | zeljkoMON | [TK]D-Fender but if i include then i will inherit some includes |
| 16:49.34 | [TK]D-Fender | Ast001: libpri, zaptel (dahdi), asteris |
| 16:49.46 | zeljkoMON | and all will be able to make all calls |
| 16:49.53 | [TK]D-Fender | zeljkoMON: then you need to make a better structure |
| 16:49.57 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:50.01 | *** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com) |
| 16:50.37 | [TK]D-Fender | zeljkoMON: [local] and [net] shound NEVER include other contexts, and no device should ever be set to use either as their context. |
| 16:50.54 | Ast001 | [TK]D-Fender what about libtermcap-devel ? Does libpri or zaptel needs it ? |
| 16:51.11 | [TK]D-Fender | Ast001: yes, IIRC |
| 16:51.36 | [TK]D-Fender | Ast001: Plenty of Ubuntu from source guides out there. Go read |
| 16:51.53 | Ast001 | ok |
| 16:53.34 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 16:53.40 | mikealeonetti | is that a no? :D |
| 16:53.53 | stevie[xxx] | how is asterisk 1.6 detecting for faxes, is NVfax still there? |
| 16:53.54 | mikealeonetti | sometimes Google doesn't want to be my friend |
| 16:54.35 | Carlos_PHX | [TK]D-Fender: Is iIRC the name for Apple's new IRC client? |
| 16:54.54 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:55.42 | zeljkoMON | hmm, yhen only way i see is to make another context that will have all users in it |
| 16:55.45 | [TK]D-Fender | Carlos_PHX: You've jsut been upgraded from dumb-ass to smart-ass. Congratulations! |
| 16:56.09 | [TK]D-Fender | zeljkoMON: Then you need to see differently. |
| 16:56.35 | [TK]D-Fender | zeljkoMON: You make a new context that combines the things you want GROUP-A to ahve access to and use THAT as the context for the device. |
| 16:56.46 | [TK]D-Fender | zeljkoMON: then a similar grouping context for OTHERS |
| 17:00.48 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 17:02.51 | zeljkoMON | [TK]D-Fender yhx, i think i can see the point |
| 17:02.54 | zeljkoMON | *thx |
| 17:03.11 | [TK]D-Fender | zeljkoMON: Excellent |
| 17:03.21 | [TK]D-Fender | zeljkoMON: This is a very important thing to learn |
| 17:03.33 | zeljkoMON | doing things right way |
| 17:11.12 | stevie[xxx] | can someone tell my the asterisk 1.6 fax_detect command? |
| 17:13.33 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:22.04 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:26.43 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 17:28.36 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:29.05 | *** join/#asterisk RobertLaptop (n=rmiddle@m9f0736d0.tmodns.net) |
| 17:29.15 | ACK-NAK | Question: Why is chan_features disabled by default in 1.6 in menuselect. Should I enable it? |
| 17:29.15 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:30.44 | file | ACK-NAK: no, you do not need it |
| 17:30.57 | ACK-NAK | file: thank you! |
| 17:31.06 | file | thus why it is disabled |
| 17:37.23 | *** join/#asterisk catch23 (n=catch23@38.113.112.17) |
| 17:38.25 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:38.33 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-4ea32c97c5e435c7) |
| 17:39.21 | catch23 | hi, i'm using asterisk as my personal line right now and every now and then I get bombarded with robocalls that last almost 3 hours, usually when I'm not around. Asterisk then creates a giant wav file and sends it to me as voicemail. Is there a way I can have asterisk hang up after a certain amount of time so I don't waste minutes? or is there a way to detect these robocalls somehow? |
| 17:42.29 | ajohnson | catch23: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 17:42.49 | ajohnson | maxmessage |
| 17:42.49 | ajohnson | This defines the maximum amount of time in seconds of an incoming message. |
| 17:43.49 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:45.00 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 17:46.23 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 17:46.49 | catch23 | ajohnson: ah cool, will asterisk hang up the call after maxmessage, or does it record the full length and then truncate the audio file? |
| 17:47.08 | ajohnson | It hangs up |
| 17:47.28 | catch23 | that's good. thanks |
| 17:51.56 | *** join/#asterisk marcrosoft (n=mark@97-119-205-119.hlna.qwest.net) |
| 17:52.10 | *** part/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:52.11 | marcrosoft | kinda a dumb question, but how do you dial an extension with xlite |
| 17:52.16 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:52.17 | marcrosoft | is it just the number? |
| 17:52.41 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 17:53.33 | Bad_Robot- | just the numberf |
| 17:53.36 | Bad_Robot- | number |
| 17:53.58 | marcrosoft | k thanks |
| 17:54.07 | ACK-NAK | catch23: Usually such calls have a toll-free or blocked callerID. You can put them in a simple IVR that requires something that a bot can't do. Press 1 if you are not a telemarketer. I get about a zillion calls a month from St Louis for some incorrectly published number. I put all 314 calls into such an IVR. |
| 17:54.32 | *** join/#asterisk scalex000 (n=chatzill@179.120.88.200.f.sta.codetel.net.do) |
| 17:54.44 | Bad_Robot- | if i got that many bogus calls i'd change my number |
| 17:54.51 | scalex000 | hello awk_r |
| 17:55.29 | awk_r | scalex000, welcome. still having dependency issues? |
| 17:56.17 | ACK-NAK | Bad_Robot-: Good idea, except my 400 year old grandmother would have learn a new number |
| 17:56.19 | scalex000 | I ask you few minute ago in wrong room about how to find temp.. |
| 17:57.53 | Bad_Robot- | just go to her house and program in a speeddial ;) |
| 17:58.43 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:58.44 | Bad_Robot- | i'm just having one of those days where i just don't want to work :( |
| 17:59.59 | scalex000 | Ok Im back |
| 18:00.33 | scalex000 | where can I found to install termcap |
| 18:00.42 | *** join/#asterisk jlnt (n=jlnt@70.255.193.190) |
| 18:01.13 | marcrosoft | is there a default extention installed for testing in asterisk. i have my softphone in registerd and it says ready |
| 18:01.33 | *** part/#asterisk packetstream (n=chatzill@77.240.56.22) |
| 18:02.10 | jlnt | unless you put something in there isn't anything |
| 18:02.41 | marcrosoft | well i put in extention 3 and 4 |
| 18:02.51 | marcrosoft | have two softphones on two computers |
| 18:03.02 | jlnt | then it should work |
| 18:03.07 | marcrosoft | exten => 3,1,Dial(SIP/Phone3,20,tr) |
| 18:03.08 | marcrosoft | exten => 4,1,Dial(SIP/Phone4,20,tr) |
| 18:03.37 | jlnt | they are not dialing each other or what? |
| 18:03.38 | marcrosoft | i get person you are calling is unavailable |
| 18:03.59 | jlnt | it's not registering then |
| 18:04.14 | marcrosoft | is that xlite's message or the pbx saying that |
| 18:04.18 | jlnt | have you checked your sip.conf |
| 18:04.24 | jlnt | pbx |
| 18:04.41 | marcrosoft | in sip.conf |
| 18:04.54 | marcrosoft | i have [Phone3] |
| 18:04.55 | *** join/#asterisk bpgoldsb (n=bpgoldsb@spatialdata2-gru-gw.customer.gru.net) |
| 18:05.07 | marcrosoft | and a couple of lines showing ip and stuff |
| 18:05.17 | marcrosoft | at the end of the file, i havnt touched anything else |
| 18:06.06 | *** join/#asterisk jer_ (n=jer@unaffiliated/jer) |
| 18:06.19 | marcrosoft | http://pastebin.com/mf388cd9 |
| 18:06.51 | marcrosoft | to dial the extention I am hitting 4 and hitting send |
| 18:06.55 | marcrosoft | is that correct? |
| 18:08.03 | *** join/#asterisk wtsexton00 (n=tim@potatosalad.worldspice.net) |
| 18:08.14 | jlnt | you have your secret= line right? |
| 18:08.25 | marcrosoft | no i have no password |
| 18:08.33 | marcrosoft | should i set it |
| 18:08.36 | catch23 | ACK-NAK: oh that's a brilliant idea... |
| 18:08.37 | scalex000 | awk_r |
| 18:08.44 | jlnt | I would try that |
| 18:08.54 | wtsexton00 | polycom keeps adding more and more junk to the firmware, anyone know how to disable background menu option? |
| 18:08.55 | scalex000 | can you help how to install asterisk prerrequisites |
| 18:09.25 | jlnt | do you have a NAT? |
| 18:09.31 | jlnt | firewall that could be blocking it as well? |
| 18:09.38 | catch23 | ACK-NAK: is there a sound file somewhere out there that plays back "if you're not a telemarketer press 1?" ... not that my own voice isn't good enough :P |
| 18:09.40 | marcrosoft | not between those computers |
| 18:09.58 | jlnt | What OS are they running? |
| 18:10.06 | marcrosoft | asterisk is on linux |
| 18:10.14 | marcrosoft | two softpones use windows |
| 18:10.21 | jlnt | no windows firewall on? |
| 18:10.32 | marcrosoft | let me double check |
| 18:10.50 | jlnt | I've seen that block them from making calls between each other before |
| 18:10.54 | awk_r | scalex000, http://www.asterisk.org/support/install here is a list of the packages (under "Build the source") |
| 18:11.46 | marcrosoft | defaultip = 192.168.90.100 |
| 18:11.50 | jlnt | I run eyebeam for my softphones |
| 18:11.53 | marcrosoft | does that have to match the computer that hosts the softphone |
| 18:11.57 | marcrosoft | or different |
| 18:12.18 | jlnt | I would temporarily disable the firewall on both machines to test |
| 18:12.30 | marcrosoft | the windows firewalls are both disabled |
| 18:12.47 | jlnt | then on your asterisk machine ssh to it then at the CLI type sip show peers |
| 18:12.52 | jlnt | and make sure they show up there |
| 18:13.01 | ACK-NAK | catch23: I'm guessing not. You may be able to stitch together some alison strings or use a text-to-speech engine. Have stephen hawking work as your personal assistant! |
| 18:13.47 | marcrosoft | jlnt: does defaultip= in sip.conf have to match the computer that has the softphone? |
| 18:13.54 | marcrosoft | jlnt: if it does that is my problem |
| 18:13.55 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:14.08 | [TK]D-Fender | marcrosoft: defaultip should not be used. let your phones register normally. |
| 18:14.15 | jlnt | yeah |
| 18:14.20 | marcrosoft | k let me take that off |
| 18:14.35 | jlnt | unless the computer is running a static ip I wouldn't use defaultip |
| 18:14.43 | [TK]D-Fender | marcrosoft: Nor should you set the IP anywhere... use "host=dynamic" |
| 18:15.00 | marcrosoft | lol |
| 18:15.04 | marcrosoft | is says host=dynamic |
| 18:15.07 | marcrosoft | and under that i have ip set |
| 18:15.08 | marcrosoft | silly |
| 18:15.11 | jlnt | also no spaces in your conf |
| 18:15.20 | jlnt | it should look like this |
| 18:15.49 | marcrosoft | do you have to restart asterisk after making .conf changes? |
| 18:16.13 | *** join/#asterisk wierdo (i=wierdo@77.78.3.107) |
| 18:16.13 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:16.14 | ACK-NAK | catch23: That's if he's not too busy recording those slicky-hella gangsta-rap tunes. |
| 18:16.38 | jlnt | I replied to your link |
| 18:16.51 | jlnt | I would |
| 18:16.55 | *** join/#asterisk lou_gr (n=lou_gr@212-70-216-131.ath.static.tee.gr) |
| 18:17.17 | wtsexton00 | dang polycom wish I could stay with 2.2, now I've got to figure out how to disable the backgrounds options |
| 18:17.43 | jlnt | what polycom phone is it? |
| 18:17.47 | jlnt | 501? |
| 18:17.51 | wtsexton00 | 560 |
| 18:17.56 | wtsexton00 | 650 I mean |
| 18:18.07 | jlnt | err don't know |
| 18:18.10 | jlnt | I have the 501's |
| 18:18.17 | wtsexton00 | can't find the dang option in the sip.cfg to disable it |
| 18:18.27 | wtsexton00 | its something new in the 3.x firmware |
| 18:18.48 | jlnt | I'll do some research for you |
| 18:18.50 | wtsexton00 | We've been using 2.2 forever but the 650 sidecar backlight won't work under 2.2 |
| 18:18.51 | jlnt | see what I can find out |
| 18:19.17 | wtsexton00 | yea thanks I can delete all the images of course or remove them line by line but it'll leave the option thistle no matter what |
| 18:20.07 | jlnt | macrosoft: did you get it figured out |
| 18:21.16 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:21.51 | marcrosoft | sip show peers shows command not found |
| 18:21.53 | marcrosoft | ruh roh |
| 18:22.06 | jlnt | did you do asterisk -r first |
| 18:22.21 | marcrosoft | oh. hehe |
| 18:22.24 | jlnt | lol |
| 18:22.30 | jlnt | it's all good I left that part out |
| 18:22.33 | marcrosoft | k it shows the 2 phones |
| 18:22.39 | jlnt | that's a + |
| 18:22.42 | marcrosoft | status unmonitored |
| 18:22.57 | marcrosoft | is that a good sign? |
| 18:23.00 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 18:23.05 | jlnt | yeah |
| 18:23.30 | marcrosoft | so why does it sday the person is unavailable :) |
| 18:24.08 | wtsexton00 | unmonitored normally means the phone/device isn't connected |
| 18:24.16 | jlnt | hmm |
| 18:24.18 | marcrosoft | oh |
| 18:24.22 | jlnt | that's the status |
| 18:24.29 | jlnt | I just ran the command to see what you were talking about |
| 18:24.35 | jlnt | did you restart your asterisk |
| 18:24.43 | jlnt | then try setting up the accounts again in x lite |
| 18:24.46 | jlnt | to see if they connect then |
| 18:25.39 | marcrosoft | i restarted then closed the softphone and opened again |
| 18:25.51 | jlnt | k, run the command sip show peers again |
| 18:25.59 | jlnt | it should show Status OK then a ping |
| 18:26.19 | root52 | Can anyone help sort throught why it seems the peer is registered but when i try to make an incomeing call on it I get 401 Unauthorized but then it looks like it registers again. Here is the sip debug peer output http://pastebin.ca/1275141 I will say it could be a problem with the provider they have been flakey in the past. Thanks for any insight. |
| 18:26.21 | jlnt | correct wtsexton00? |
| 18:26.38 | wtsexton00 | yea |
| 18:26.40 | wtsexton00 | something like |
| 18:26.45 | marcrosoft | send outbout via? what should that be set to? |
| 18:26.47 | wtsexton00 | 207 (Unspecified) D 0 UNKNOWN |
| 18:26.47 | wtsexton00 | 206/206 192.168.1.62 D 5060 OK (14 ms) |
| 18:27.00 | marcrosoft | im reconfiguring the softphone to be sure |
| 18:27.14 | jlnt | astwatch: Bad exit status from `/usr/bin/pgrep -f connecting > /dev/null && /usr/bin/kill -9 `/usr/bin/pgrep -f connecting``: 9 |
| 18:27.14 | jlnt | <PROTECTED> |
| 18:27.15 | jlnt | lol |
| 18:27.18 | wtsexton00 | shows 207 isn't registered and 206 is |
| 18:27.46 | wtsexton00 | I haven't |
| 18:27.55 | jlnt | kicks machine :-D |
| 18:28.12 | marcrosoft | still shows unmonitored |
| 18:28.16 | jlnt | I'll just take it out of astwatch |
| 18:28.34 | jlnt | hmm |
| 18:28.40 | jlnt | and your softphone shows ready |
| 18:28.47 | marcrosoft | yes |
| 18:29.04 | marcrosoft | and if i change the domain ip in the softphone it wont say ready |
| 18:29.11 | marcrosoft | so it is definitely connecting |
| 18:29.36 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 18:29.40 | ghento | Hi all. I'm attempting to use Read(), with maxdigits set to 1. However when I input a digit on my GSM phone nothing is being read? |
| 18:29.42 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:30.06 | jlnt | I am unsure |
| 18:30.33 | Micc | what does it mean when I show channels and it says AppQueue((Outgoing Line)) |
| 18:30.48 | *** join/#asterisk lord_nikon (n=lord@host-216-153-131-74.roc.choiceone.net) |
| 18:30.58 | marcrosoft | Both firewalls are off |
| 18:31.50 | jlnt | Micc: it shows active calls |
| 18:31.52 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 18:32.19 | jlnt | I am doing some research now macrosoft |
| 18:32.35 | lord_nikon | im having an issue trying to transfer calls from one * server to another, i keep getting 407 requests from the destination. is there any way i can disable that ? |
| 18:33.06 | ruben23 | hi |
| 18:33.08 | marcrosoft | jlnt: this site says unmonitored doesnt mean it isnt on |
| 18:33.14 | marcrosoft | you have to say, qualify=yes |
| 18:33.21 | marcrosoft | let me add that |
| 18:33.42 | jlnt | I'm looking at my conf file |
| 18:33.48 | jlnt | I may just post an example |
| 18:34.16 | wtsexton00 | eh I'm beginning to not like polycom |
| 18:34.19 | ruben23 | how to install mysql client on my asterisk boxes...i tried this link not found...http://mirror.trouble-free.net/mysql_mirror/Downloads/MySQL-4.0/mysql-4.0.27.tar.gz |
| 18:34.23 | jlnt | qualify=2000 |
| 18:34.33 | jlnt | pickupgroup=1 |
| 18:34.36 | jlnt | callgroup=1 |
| 18:34.41 | jlnt | context=internal |
| 18:34.54 | jlnt | that's a few extra lines on mine |
| 18:35.04 | marcrosoft | k let me try that |
| 18:35.07 | hardwire | anybody use logrotate vs 'logger rotate' for asterisk logs? |
| 18:35.29 | [TK]D-Fender | Polycom > All |
| 18:35.33 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 18:36.00 | ruben23 | anyone have ideas.. |
| 18:36.02 | Micc | hardwire, I've had problems with logger rotate in asterisk, but only because I hit the file system limit for a file size. |
| 18:36.22 | hardwire | oh, I do as well |
| 18:36.25 | hardwire | rotating more often now |
| 18:36.37 | Micc | hardwire, the file was a sql log file, but it still caused logger rotate to go crazy and create thousands of files. |
| 18:36.43 | marcrosoft | still says unmonitored |
| 18:36.46 | marcrosoft | :( |
| 18:36.49 | hardwire | Micc: ooh, interesting |
| 18:36.57 | *** join/#asterisk zxd (n=zapw@213.31.43.2) |
| 18:36.58 | zxd | hello |
| 18:37.02 | Micc | hardwire, I trust it does its job as long as it doesn't run into any problems. |
| 18:37.07 | jlnt | did you restart marcrosoft |
| 18:37.09 | [TK]D-Fender | marcrosoft: pastebin your new config and sip peers dump |
| 18:37.12 | marcrosoft | yes |
| 18:37.25 | JonOnt | Hey guys, whats the best codec for fax over sip, we dont have T.38 handoff? |
| 18:37.31 | zxd | is it possible to configure the tos byte for rtcp packets in asterisk / |
| 18:37.32 | zxd | ? |
| 18:37.40 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 18:37.50 | [TK]D-Fender | JonOnt: the ONLY real option is G.711 |
| 18:38.08 | zxd | i see tos_audio , under sip.conf , but that dosen't seem to do anything to rtcp packets and the rtp data |
| 18:38.12 | marcrosoft | http://pastebin.com/m51b88177 |
| 18:38.16 | zxd | i mean only the rtp data |
| 18:38.29 | zxd | rtcp packets dscp value isn't set to that value |
| 18:38.29 | Micc | hardwire, you might not ever run into that problem if you just use logrotate because it would run externaly. But you may still have asterisk problems and not know it till its too late. |
| 18:38.35 | [TK]D-Fender | [13:33]<marcrosoft>you have to say, qualify=yes |
| 18:38.45 | [TK]D-Fender | marcrosoft: unload res_dyslexia.so |
| 18:38.55 | hardwire | Micc: yeh, that's what I'm trying to figure out.. |
| 18:38.56 | marcrosoft | lol |
| 18:39.17 | marcrosoft | well i tried qualify=yes before qualify=2000... let me try it again |
| 18:39.37 | jlnt | lol |
| 18:39.45 | jlnt | quality=2000 |
| 18:39.48 | jlnt | qualify=yes |
| 18:40.19 | jlnt | nvm |
| 18:40.22 | jlnt | cancel quality |
| 18:40.35 | jlnt | mine does say qualify |
| 18:40.42 | jlnt | but then again this is a fonality system |
| 18:40.51 | root52 | From Above Neverminde must have just been the flakey provider again because it is all working now. |
| 18:40.54 | marcrosoft | well it says unmonitored still |
| 18:40.57 | marcrosoft | and it doesnt work |
| 18:40.59 | marcrosoft | :( |
| 18:41.25 | jlnt | that's good root |
| 18:41.42 | jlnt | sucks about the provider though |
| 18:42.01 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:42.16 | root52 | yeah well lets just say you get what you pay for. It is a hoby system so I try no to shell out to much cash ;-) |
| 18:42.38 | jlnt | lol |
| 18:42.40 | jlnt | that's true |
| 18:42.42 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.201.204) |
| 18:43.11 | [TK]D-Fender | [13:39]<marcrosoft>well i tried qualify=yes before qualify=2000... let me try it again |
| 18:43.26 | [TK]D-Fender | marcrosoft: http://pastebin.com/m51b88177 |
| 18:43.39 | [TK]D-Fender | marcrosoft: do YOU see the word "QUALIFY" in there? I know *I* don't |
| 18:43.45 | [TK]D-Fender | marcrosoft: unload res_dyslexia.so |
| 18:43.53 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.static.ip.windstream.net) |
| 18:44.17 | marcrosoft | [TK]D-Fender: k, sec |
| 18:44.20 | wtsexton00 | one letter off ftl |
| 18:44.20 | *** part/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.static.ip.windstream.net) |
| 18:44.32 | jlnt | lol |
| 18:44.40 | [TK]D-Fender | I'm worndering how much more blatant I can be... |
| 18:44.50 | [TK]D-Fender | goes to look for his giant flashing neon sign.... |
| 18:45.00 | jlnt | hehehe |
| 18:45.09 | [TK]D-Fender | damn... no extension cord... |
| 18:45.10 | jlnt | yeah it says QUALITY not QUALIFY |
| 18:45.12 | wtsexton00 | if I can figure out how to get rid of this âThistleâ I'll be good |
| 18:45.16 | marcrosoft | lol |
| 18:45.41 | marcrosoft | alright it says ok |
| 18:45.45 | marcrosoft | 106ms |
| 18:45.49 | jlnt | you can make calls now |
| 18:45.52 | jlnt | that's awesome man |
| 18:45.52 | wtsexton00 | don't want people putting images on their damned phones |
| 18:45.53 | jlnt | try it |
| 18:46.03 | jlnt | why not wt |
| 18:46.15 | wtsexton00 | a 650 with three side cars is retarded looking to begin with |
| 18:46.25 | jlnt | HAHAHAHAHAHQA |
| 18:46.28 | wtsexton00 | jlnt, well if you give them the option of flowers, they'll want cars |
| 18:46.35 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 18:46.45 | wtsexton00 | then someone will want a picture of their kid, its best to just not allow it period |
| 18:46.53 | jlnt | yeah |
| 18:47.04 | wtsexton00 | we don't support custom ring tones either, you use the sucky tones polycom gives us or go without |
| 18:47.05 | marcrosoft | sigh... still getting the same voice recording |
| 18:47.25 | jlnt | and they are showing up now |
| 18:47.26 | marcrosoft | maybe softphone settings are not correct |
| 18:47.28 | jlnt | not as unknown |
| 18:47.42 | wtsexton00 | you'll want to take a look at your dial plan and your console now |
| 18:47.49 | marcrosoft | person is unavailable... please try again |
| 18:47.59 | jlnt | unclick DND |
| 18:48.00 | jlnt | haha |
| 18:48.17 | jlnt | yeah let me lookup the dial plan for you |
| 18:48.32 | jlnt | or paste your extensions.conf |
| 18:48.33 | wtsexton00 | eh, I'll just take write access to the mac-phone.cfg :) |
| 18:48.37 | marcrosoft | jlnt: i clicked dnd on and off.. no workie |
| 18:48.44 | [TK]D-Fender | marcrosoft: PB your extensions.conf |
| 18:48.48 | jlnt | yeah, I was just kidding |
| 18:48.57 | marcrosoft | [TK]D-Fender: the whole thing or just the part i modified? |
| 18:49.02 | jlnt | whole |
| 18:49.04 | [TK]D-Fender | marcrosoft: WHOE THING |
| 18:49.05 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 18:49.20 | [TK]D-Fender | marcrosoft: I'm not going to trust little bits at a time. Hope you understand :) |
| 18:49.35 | wtsexton00 | I'm sure polycom will want a support contract if I email them on how to disable it |
| 18:49.42 | marcrosoft | [TK]D-Fender: lol, well with the quatity qualify i understand :P |
| 18:50.06 | [TK]D-Fender | marcrosoft: Thats only for the fact I had to tell you twice :p |
| 18:52.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:52.56 | marcrosoft | [TK]D-Fender: http://pastebin.com/f8ebe93 |
| 18:52.59 | *** join/#asterisk bruns8234 (n=chatzill@p5B08F163.dip.t-dialin.net) |
| 18:54.03 | [TK]D-Fender | marcrosoft: Yup... |
| 18:54.12 | [TK]D-Fender | marcrosoft: So... what context are your phones to sue? |
| 18:54.15 | [TK]D-Fender | use* |
| 18:54.30 | [TK]D-Fender | marcrosoft: And where did you put those 2 extens you made? |
| 18:54.42 | [TK]D-Fender | marcrosoft: LMK when it hits you :) |
| 18:56.28 | marcrosoft | [TK]D-Fender: should it be in local? |
| 18:56.47 | *** part/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 18:57.02 | jlnt | are you talking about on the file system? |
| 18:57.07 | [TK]D-Fender | marcrosoft: where did you TELL your phones to send their calls? |
| 18:57.13 | jlnt | both files should be /etc/asterisk |
| 18:57.25 | marcrosoft | [TK]D-Fender: to the ip of the asterisk server |
| 18:57.36 | [TK]D-Fender | marcrosoft: NO. |
| 18:57.37 | jlnt | with what port |
| 18:57.40 | jlnt | lol |
| 18:57.53 | [TK]D-Fender | marcrosoft: Your sip.conf tells * what CONTEXT to send the calls it gets from them. |
| 18:58.23 | marcrosoft | ok, those were Phone3 and Phone4 |
| 18:58.52 | jlnt | at the very bottom |
| 18:59.18 | marcrosoft | jlnt: yea the extens i added where at the bottom |
| 18:59.30 | jlnt | I see them |
| 18:59.36 | [TK]D-Fender | marcrosoft: read again. |
| 19:00.03 | [TK]D-Fender | marcrosoft: sip.conf tells what context in the DIALPLAN * will send calls from your devices. |
| 19:00.16 | [TK]D-Fender | marcrosoft: what context is it sending them to? |
| 19:00.29 | marcrosoft | Phone3, Phone4 |
| 19:00.31 | [TK]D-Fender | NO |
| 19:00.37 | [TK]D-Fender | marcrosoft: those are DEVICE NAMES. |
| 19:01.00 | marcrosoft | maybe it is in authentication |
| 19:01.03 | marcrosoft | [authentication] |
| 19:01.11 | unpaidbill | wow, 1.6 fixed my 7935 with skinny, this is good news. |
| 19:01.19 | unpaidbill | high five dudes. |
| 19:01.33 | marcrosoft | [TK]D-Fender: Phone3, Phone4 are at the bottom of the sip.conf |
| 19:01.45 | marcrosoft | [TK]D-Fender: maybe should they be placed in general? |
| 19:02.11 | marcrosoft | [TK]D-Fender: or am i way off |
| 19:03.32 | marcrosoft | [TK]D-Fender: context is default |
| 19:04.04 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:04.51 | marcrosoft | and those two exten lines are in [default]... so i guess i don't get it |
| 19:05.30 | [TK]D-Fender | marcrosoft: [phone4} is NOT A CONTEXT. it is the DEVICE NAME |
| 19:05.49 | [TK]D-Fender | marcrosoft: what CONTEXT did you tell your DEVICES to send their calls to? |
| 19:06.14 | marcrosoft | [TK]D-Fender: sip |
| 19:06.22 | [TK]D-Fender | marcrosoft: you did it with these magical little things called PARAMETERS you set below the device name |
| 19:06.26 | marcrosoft | [TK]D-Fender: which doesnt exist probable in extentions.conf |
| 19:06.40 | marcrosoft | [TK]D-Fender: so change that do default or something |
| 19:06.42 | [TK]D-Fender | marcrosoft: And what is in this [sip] context in extensions.conf? |
| 19:06.55 | marcrosoft | [TK]D-Fender: ill check, but i doubt it exists |
| 19:07.03 | [TK]D-Fender | marcrosoft: I don't |
| 19:07.44 | marcrosoft | [TK]D-Fender: no sip context |
| 19:08.03 | [TK]D-Fender | marcrosoft: So you have pointed your phones to a dead end. hence they can dial nothing |
| 19:08.16 | [TK]D-Fender | marcrosoft: another lesson down... |
| 19:08.36 | marcrosoft | [TK]D-Fender: thanks for the help |
| 19:08.40 | marcrosoft | [TK]D-Fender: and the patience |
| 19:09.02 | [TK]D-Fender | marcrosoft: healthy starting tip. trash your extensions.conf except for the [general] and [globals] contexts |
| 19:09.15 | [TK]D-Fender | marcrosoft: And stuff you did yourself. |
| 19:09.22 | marcrosoft | holy crap it's dialing :P |
| 19:09.39 | [TK]D-Fender | marcrosoft: And remove all the comments. |
| 19:09.41 | marcrosoft | [TK]D-Fender: will do |
| 19:10.04 | [TK]D-Fender | marcrosoft: When you start to learn * its best to build it yourself from scratch without crap in the way |
| 19:10.15 | marcrosoft | [TK]D-Fender: makes sense |
| 19:10.23 | marcrosoft | [TK]D-Fender: at least now i have something to go with |
| 19:10.28 | [TK]D-Fender | marcrosoft: this isn't like Apache where a ton of stuff is actually of some value starting from the samples./ |
| 19:10.35 | marcrosoft | [TK]D-Fender: would have probable given up if i couldnt even get a softphone going |
| 19:10.48 | marcrosoft | probably |
| 19:11.03 | jlnt | when you setting up trunks |
| 19:11.10 | [TK]D-Fender | marcrosoft: *'s samples are for reference only... (the core stuff anyways) |
| 19:11.34 | [TK]D-Fender | marcrosoft: asterisk.conf, modules.confand a bunch of others are fine, but sip.conf and extensions.conf have got to be trashed |
| 19:11.46 | [TK]D-Fender | marcrosoft: or your learning experience will be cluttered with crap |
| 19:11.48 | espent | is there any way to run some script after a user is left (hung up) from a agi/dialplan? |
| 19:12.05 | [TK]D-Fender | espent: "h" standa extension |
| 19:12.08 | [TK]D-Fender | standard |
| 19:12.28 | espent | [TK]D-Fender: does that go for calls created by originate from AMI? |
| 19:12.40 | marcrosoft | [TK]D-Fender: so basically if i had a sip phone i could just hook it to the network, configure it the same way and have a real phone to ring |
| 19:12.46 | [TK]D-Fender | espent: A cal is a call is a call |
| 19:13.09 | [TK]D-Fender | marcrosoft: if you want to call that a "real" phone, sure. |
| 19:13.22 | [TK]D-Fender | marcrosoft: certainly more so than a soft-phone |
| 19:13.42 | marcrosoft | [TK]D-Fender: well it is a local phone i guess |
| 19:13.47 | [TK]D-Fender | marcrosoft: So... go trash EVERYTHING you didn't do yourslef and pastebin your 2 new configs |
| 19:14.08 | [TK]D-Fender | marcrosoft: depends on your definition of "local" as well :) |
| 19:14.17 | espent | [TK]D-Fender: thanks a lot, that'll keep me going :) |
| 19:14.34 | jlnt | anyone know a good VOIP provider? |
| 19:14.45 | jaytee | larry |
| 19:14.51 | *** join/#asterisk sprite-- (n=sprite@12.228.1.97) |
| 19:15.10 | sprite-- | What's the best/easiest way to upgrade my AsteriskNOW1.5b box to use Asterisk 1.6 instead of 1.4? |
| 19:16.15 | *** part/#asterisk synchris (n=synchris@athedsl-4381731.home.otenet.gr) |
| 19:16.55 | [TK]D-Fender | sprite--: its available in their repo IIRC |
| 19:17.51 | marcrosoft | [TK]D-Fender: keep general and globals in sip.conf as well? |
| 19:18.00 | sprite-- | so yum remove asterisk14 and yum install asterisk16? |
| 19:18.14 | [TK]D-Fender | marcrosoft: Yes, trashing everytihng that is commented out. |
| 19:18.31 | [TK]D-Fender | sprite--: pretyy much how it worlks |
| 19:19.09 | sprite-- | [TK]D-Fender: Thanks, didn't know if that was going to remove my config files or not, if it was the proper way to ugprade |
| 19:19.33 | [TK]D-Fender | sprite--: I'd back those up if I were you |
| 19:20.59 | *** join/#asterisk Leddy (n=Leddy@72.54.198.194) |
| 19:21.44 | marcrosoft | [TK]D-Fender: http://pastebin.com/f596ea35c , http://pastebin.com/f34f826ae |
| 19:24.44 | *** join/#asterisk pecanha (n=e@189.106.43.63) |
| 19:26.12 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-fa9c7b7d0948f5f6) |
| 19:26.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:28.57 | [TK]D-Fender | marcrosoft: Excellent |
| 19:29.09 | [TK]D-Fender | marcrosoft: Now you can get started. |
| 19:29.23 | marcrosoft | [TK]D-Fender: got any suggested reading for homework :) |
| 19:29.36 | [TK]D-Fender | marcrosoft: ... |
| 19:29.37 | [TK]D-Fender | ~book |
| 19:29.38 | jbot | i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:29.49 | [TK]D-Fender | marcrosoft: and for a small sample config for some inspiration : |
| 19:29.51 | [TK]D-Fender | ~jerjerguide |
| 19:29.52 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 19:30.09 | pecanha | Hello! Could anyone with xlite and not behind NAT connect to my extension number and just help me to figure out if the problem is not really the NAT?? |
| 19:30.54 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 19:31.17 | marcrosoft | [TK]D-Fender: nice... is that your blog? |
| 19:31.20 | [TK]D-Fender | pecanha: pastebin syour sip.conf masking only passwords. |
| 19:31.36 | [TK]D-Fender | marcrosoft: No, but I was a key contributor to that blog entry |
| 19:31.42 | [TK]D-Fender | ~pb |
| 19:31.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 19:31.44 | [TK]D-Fender | ^^^^^^^^ |
| 19:32.12 | pecanha | [TK]D-Fender: my sip conf includes few files, so I'll put the main includes, ok? |
| 19:32.26 | marcrosoft | I might add, pastebinit is a linux package that will auto pastebin stuff for you... |
| 19:32.47 | marcrosoft | pastebinit somefile.conf |
| 19:33.20 | [TK]D-Fender | pecanha: Yes |
| 19:33.31 | [TK]D-Fender | pecEVERYTHING involved |
| 19:36.36 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:37.11 | pecanha | [TK]D-Fender: http://pastebin.ca/1275198 |
| 19:37.23 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:37.55 | [TK]D-Fender | pecanha: Sure doesn't look like "everything" to me. And is * behind NAT as well? |
| 19:39.27 | pecanha | [TK]D-Fender: My local network is behind nat, and my server is bound with a real IP, but I even receive logs from calls when using nat=no |
| 19:39.57 | pecanha | [TK]D-Fender: already tried putting my softphone machine on DMZ, opened ports etc... |
| 19:40.33 | [TK]D-Fender | pecanha: if * has a public IP, then each of your remote peers should have "nat=yes" , "qualify=yes", "canreinvite=no" |
| 19:40.47 | [TK]D-Fender | pecanha: No DMZ or forwarding necessary |
| 19:41.08 | *** join/#asterisk SiberAIR (n=SibRphre@rrcs-24-39-122-73.nyc.biz.rr.com) |
| 19:41.22 | scalex000 | hello |
| 19:41.45 | scalex000 | I need help about termcap where I can find to install on fedora |
| 19:41.49 | pecanha | my server has a public ip, my softphone not, so I should use nat=yes, qualify=yes and careinvite=no right |
| 19:42.05 | [TK]D-Fender | scalex000: yum install libtermcap |
| 19:42.17 | [TK]D-Fender | pecanha: as I said |
| 19:42.22 | scalex000 | ok |
| 19:42.24 | scalex000 | thanks |
| 19:43.36 | pecanha | anything else? cause I still get the same error |
| 19:43.38 | [TK]D-Fender | pecanha: enable sip debug and pastebint he complete failed attempt |
| 19:43.44 | pecanha | ok |
| 19:45.27 | pecanha | [TK]D-Fender: http://pastebin.ca/1275206 |
| 19:47.22 | [TK]D-Fender | pecanha: Your target phone has no codecs specified |
| 19:47.27 | sprite-- | [TK]D-Fender: When installing it through yum how do I install asterisk as user asterisk instead of root? |
| 19:47.52 | [TK]D-Fender | sprite--: AFAIK the package has its own rules |
| 19:48.05 | [TK]D-Fender | sprite--: and I'm sure it runs as non-root |
| 19:48.18 | pecanha | [TK]D-Fender: softphone? |
| 19:48.36 | [TK]D-Fender | pecanha: [1001] |
| 19:49.05 | pecanha | hmm, how can I fix ? |
| 19:49.24 | *** join/#asterisk Segnale007 (n=Pietro@host71-242-dynamic.9-79-r.retail.telecomitalia.it) |
| 19:49.26 | [TK]D-Fender | pecanha: go set them |
| 19:50.15 | pecanha | [TK]D-Fender: using disallow... allow=ilbc, right? |
| 19:50.32 | [TK]D-Fender | pecanha: like any other device |
| 19:51.02 | pecanha | [TK]D-Fender: yeah! |
| 19:51.10 | pecanha | it worked !! finally :/ |
| 19:51.28 | pecanha | thanks vry much |
| 19:52.40 | pecanha | [TK]D-Fender: u figure it out looking for config or by looking debug log? |
| 19:53.37 | [TK]D-Fender | pecanha: Your debug told me * didn't even try. Your sip peer dump showed that it knew the IP to send to. That says * already knows your peers aren't compatible so it isn't even trying |
| 19:53.43 | [TK]D-Fender | pecanha: 3 piece |
| 19:53.47 | [TK]D-Fender | (s) |
| 19:53.50 | *** join/#asterisk gpowers (n=glenn@adsl-99-142-75-162.dsl.emhril.sbcglobal.net) |
| 19:54.25 | pecanha | tks |
| 19:54.42 | [TK]D-Fender | pecanha: you're welcome |
| 20:01.47 | marcrosoft | [TK]D-Fender: do you play guitar? |
| 20:03.02 | [TK]D-Fender | marcrosoft: Yes, but has nothing to do with the origins of my nic. |
| 20:03.16 | sprite-- | I'm slightly confused, I installed Asterisk1.6 and the sip commands in the CLI are no longer there. |
| 20:03.42 | [TK]D-Fender | sprite--: probably means chan_sip didn't load. |
| 20:03.55 | [TK]D-Fender | sprite--: Forget a couple of key configs by any chance? |
| 20:04.07 | [TK]D-Fender | sprite--: Like oh I don't know.... modules.conf? asterisk.conf? |
| 20:05.55 | sprite-- | [TK]D-Fender: Hmm the modules seem to not have installed with the package. |
| 20:06.19 | [TK]D-Fender | sprite--: You'd almost think that configs were in a separate package or something! |
| 20:06.44 | sprite-- | [TK]D-Fender: I have the configs :) Not the modules |
| 20:07.10 | sprite-- | /usr/lib/asterisk/modules only has my addon modules |
| 20:14.16 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 20:14.50 | [TK]D-Fender | sprite--: Kepp shopping... |
| 20:15.07 | scalex000 | asterisk need a specific version of libtermcap |
| 20:15.22 | scalex000 | i get a error when i make ./configure |
| 20:15.28 | scalex000 | termcap un support |
| 20:17.53 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 20:19.25 | *** join/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 20:20.55 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 20:21.18 | marcrosoft | [TK]D-Fender: can you recommend me a cheap sip phone? |
| 20:22.11 | meuserj | Ok.. two things with app_voicemail_imap.so.... First: Is there any way to tell the system to revert to using local file storage if the imap folder fails for some reason (user not defined, mailbox no created, etc.) as it is now, the message is just lost. |
| 20:27.21 | sprite-- | http://rafb.net/p/L0pzYF99.html |
| 20:27.33 | sprite-- | Everything seems to be working now from the CLI, but when I call my number I get no sound. |
| 20:29.45 | Micc | what the hell is this? unable to open iax timing device |
| 20:29.58 | Micc | timing interface to be more precise |
| 20:30.06 | Micc | no such file or directory |
| 20:30.24 | Micc | everything was working fine, now my iax is not working. |
| 20:30.26 | meuserj | The second thing is probably a bit easier.. just can't find a valid config option. when the imap based voicemailbox works fine, I'm ending up with two copies of the voicemail in my e-mail. One that is dropped in through IMAP, one that is sent to me through SMTP. |
| 20:33.28 | [TK]D-Fender | marcrosoft: What do you really want to do? |
| 20:33.53 | [TK]D-Fender | Micc: says you have no zaptel/dahdi timing source and are trying to use IAX2 Trunk mode |
| 20:34.35 | [TK]D-Fender | sprite--: -- Executing [17772784063@from-pstn:1] Set("SIP/66.193.176.35-0875c9a8", "__FROM_DID=17772784063") in new stack |
| 20:35.03 | [TK]D-Fender | sprite--: Seems to say your call is coming in un-auth'd. Also are you running * behind NAT? |
| 20:35.25 | sprite-- | I am running behind a nat, I have my sip_nat.conf set up properly though. |
| 20:35.35 | sprite-- | Nothing has changed except the upgrade from 1.4->1.6 |
| 20:35.40 | [TK]D-Fender | sprite--: And the reason I trust that is...? |
| 20:36.36 | sprite-- | sip_nat.conf : nat = yes; externip = 12.228.1.97; localnet = 192.168.1.0/255.255.255.0 |
| 20:36.37 | marcrosoft | [TK]D-Fender: sorry for the delay, I want to eventually replace our phone company or at the very least replace our old pbx with some newer features and learn asterisk at the same time |
| 20:37.21 | sprite-- | I have ports 10000-20000 forwarded to the Asterisk box. Calling in from X-Lite works though, but calling in to the DID through my cell phone no longer has a voice signal. |
| 20:38.11 | scalex000 | hi I have one more question |
| 20:38.19 | scalex000 | who can help me? |
| 20:39.46 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 20:41.42 | Micc | TKD-Fender, I was able to use it before. How do I get an timing source? I thought I install zaptel dummy |
| 20:41.49 | *** part/#asterisk bruns8234 (n=chatzill@p5B08F163.dip.t-dialin.net) |
| 20:41.57 | Micc | do I need to add a load z_dummo.so or something? |
| 20:44.53 | Micc | how do I install the zaptel dummy timer? |
| 20:45.32 | Micc | why was this working just a few minutes ago? iax2 channels worked fine. |
| 20:45.38 | Micc | I was just changing extensions. |
| 20:45.50 | Micc | this is a production system. |
| 20:47.03 | *** join/#asterisk pecanha (n=e@189.106.43.63) |
| 20:47.29 | scalex000 | hello I have problem to install Asterisk. I need to install ncurse, etc |
| 20:47.37 | scalex000 | but at end said nothing to do |
| 20:48.32 | Micc | unable to create zap channel no route to host |
| 20:48.39 | Micc | or to destination. |
| 20:48.48 | Micc | WTF! this is going to drive me insane. |
| 20:48.49 | scalex000 | :( |
| 20:50.07 | [TK]D-Fender | marcrosoft: Ok, so to replace an existing company PBX? |
| 20:50.39 | marcrosoft | [TK]D-Fender: correct |
| 20:50.48 | marcrosoft | [TK]D-Fender: we have phones already but they are the older kind |
| 20:51.00 | marcrosoft | [TK]D-Fender: maybe i could use one of those converters |
| 20:51.18 | [TK]D-Fender | marcrosoft: what kind currently? |
| 20:51.36 | marcrosoft | [TK]D-Fender: lucent Partner 18D |
| 20:51.50 | marcrosoft | [TK]D-Fender: it has a standard phone line connection going o it |
| 20:51.52 | [TK]D-Fender | marcrosoft: How many? |
| 20:52.02 | [TK]D-Fender | (phones) |
| 20:52.14 | marcrosoft | [TK]D-Fender: we need like, 4-5 phones.. we have at least 10 in total laying around |
| 20:52.59 | scalex000 | hello marcrosoft |
| 20:53.12 | marcrosoft | scalex000: hello |
| 20:53.18 | [TK]D-Fender | marcrosoft: Computers already at each station? |
| 20:53.23 | marcrosoft | [TK]D-Fender: yes |
| 20:53.34 | scalex000 | I try to install asterisk on fedora 6 |
| 20:53.44 | [TK]D-Fender | marcrosoft: Ok, Polycom IP 330 for your regular users, IP 650 for your receptionist. |
| 20:53.49 | [TK]D-Fender | marcrosoft: www.telephonydepot.com |
| 20:53.57 | scalex000 | but when I type make menuselect said i need ot install ncurse, newt |
| 20:54.15 | scalex000 | when I use yum install ... finally said nothing to do |
| 20:54.18 | scalex000 | why? |
| 20:54.37 | marcrosoft | Im not familiar with fedora |
| 20:54.43 | marcrosoft | and today is my first day with asterisk |
| 20:54.49 | marcrosoft | i did manage to get it working however |
| 20:55.04 | marcrosoft | [TK]D-Fender: thanks, ill check those out.. can we utilize these older phones or just go new? |
| 20:55.15 | scalex000 | ok |
| 20:55.17 | [TK]D-Fender | marcrosoft: Forget the old phones... |
| 20:55.22 | [TK]D-Fender | marcrosoft: Dead issue |
| 20:55.22 | marcrosoft | [TK]D-Fender: ok sounds good |
| 20:55.34 | scalex000 | thanks |
| 20:55.43 | marcrosoft | scalex000: did you install ncurse? |
| 20:55.44 | stevie[xxx] | i want to pass the DID to iaxmodem extension, anyone can help here? http://nopaste.org/p/aD5bUsIBC |
| 20:56.11 | scalex000 | I try to install but said "nothing to do" I dont know why? |
| 20:56.22 | marcrosoft | scalex000: maybe it is installed |
| 20:56.44 | marcrosoft | scalex000: ive only breifly used yum/fedora |
| 20:56.51 | marcrosoft | scalex000: im more of a debian kinda guy |
| 20:56.57 | scalex000 | ok |
| 20:56.59 | scalex000 | ok |
| 20:57.14 | [TK]D-Fender | stevie[xxx]: Trixbox is not supported here. |
| 20:57.34 | sprite-- | [TK]D-Fender: Reverting to 1.4 to see if that fixes things. |
| 20:57.52 | stevie[xxx] | hmpf |
| 20:58.03 | SkramX | so. I have a queue with dynamic agents- `queueaddmember(queue,channel)` except now, I want to keep track of who authenticated for what agent/channel. Does asteriskprovide variables for this or must i do this in my own DB? |
| 20:58.32 | Micc | please someone tell me how to setup a timer for iax |
| 20:59.01 | pecanha | [TK]D-Fender: I trying now to use a linksys pap2 instead of x-lite. I'm getting the same error as before, so its probably codec again. To add more than one codec, can I use allow=ilbc&g279a as example |
| 20:59.06 | Qwell | Micc: elaborate |
| 20:59.50 | [TK]D-Fender | pecanha: First the PAP2 does not SUPPORT iLBC, and * cannot transcode to G.729 without a licenced codec. |
| 20:59.52 | sprite-- | [TK]D-Fender: Reverted to 1.4 and I have sound again when calling in from my cellphone. |
| 21:00.16 | [TK]D-Fender | Micc: Go setup ztdummy / dhadi_dummy |
| 21:00.23 | pecanha | [TK]D-Fender: hmm, which codec do you recommend for pap2? |
| 21:00.46 | [TK]D-Fender | pecanha: ULAW/ALAW depending where you are |
| 21:01.34 | pecanha | [TK]D-Fender: pap2 supports g723, g711u and g711a, g726-x |
| 21:02.00 | jasonwoot | <PROTECTED> |
| 21:02.06 | [TK]D-Fender | pecanha: g711u and g711a <-- ulaw / alaw |
| 21:02.13 | pecanha | [TK]D-Fender: ah! |
| 21:02.23 | [TK]D-Fender | jasonwoot: "netstat -an|grep 5060 |
| 21:03.47 | Micc | I'm in /usr/src/asterisk/zaptel-1.4.12.1, I've build everything. ready doesn't say how to install ztdummy |
| 21:04.06 | jasonwoot | sonofa... ty Fender |
| 21:04.53 | [TK]D-Fender | Micc: .. |
| 21:04.55 | [TK]D-Fender | ~book |
| 21:04.56 | jbot | rumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:05.01 | [TK]D-Fender | ~wikis |
| 21:05.02 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:05.21 | [TK]D-Fender | Micc: modproble ztdummy , ztcfg -vvvv |
| 21:05.37 | [TK]D-Fender | Micc: the RECOMPILE & INSTALL * |
| 21:05.52 | pecanha | What's the meaning of SIP/1001-098f7af8 is circuit-busy? sry for many questions! heh |
| 21:06.58 | [TK]D-Fender | pecanha: means "pastebin the complete call attempt's CLI output with SIP debug" |
| 21:07.16 | stevie[xxx] | anyway, is this a good way to present my problem? [TK]D-Fender, like i did this time with nopaste |
| 21:08.19 | Micc | module ztdummy not found |
| 21:08.40 | [TK]D-Fender | Micc: did a full make && make install, and rebooted? |
| 21:08.43 | Micc | I've got the ztdummy.ko file built. |
| 21:08.48 | Micc | not reboot. |
| 21:09.25 | [TK]D-Fender | Micc: You may have to use insmod depending on your distro,e tc |
| 21:09.33 | [TK]D-Fender | Micc: but I'd recommend a restart |
| 21:09.34 | Micc | CentOs 5 |
| 21:09.39 | Micc | rebooting now. |
| 21:09.58 | [TK]D-Fender | Micc: Just reboot. |
| 21:10.15 | Micc | ok. seems my ssh sessions sometimes lock up on my every so often toon. |
| 21:10.17 | Micc | ok. seems my ssh sessions sometimes lock up on my every so often too. |
| 21:11.10 | Micc | ok, still the same thing. module not found. |
| 21:13.20 | pecanha | [TK]D-Fender: is it possible to redirect output results? as redirecting to a file? |
| 21:13.38 | pecanha | can't see all debug |
| 21:13.52 | Micc | do I need to rebuild my kernel? |
| 21:14.06 | [TK]D-Fender | Micc: No. |
| 21:14.16 | [TK]D-Fender | Micc: Did you rebuild * from SCRATCH and reinstall? |
| 21:14.33 | pecanha | http://pastebin.ca/1275302 |
| 21:15.36 | [TK]D-Fender | pecanha: PB your 1001 peer |
| 21:16.29 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 21:17.16 | jasonwoot | [TK]D-Fender: where does asterisk define what interfaces it's bound to? |
| 21:18.23 | Micc | TKD-Fender, yes as far as I know. I fllowed all the instructions on a couple different sites about installing asterisk on centos 5. |
| 21:18.26 | pecanha | http://pastebin.ca/1275307 |
| 21:19.07 | [TK]D-Fender | jasonwoot: in each channel config file |
| 21:19.36 | [TK]D-Fender | Micc: As far as you know? I jsut asked you an extremely specific question. |
| 21:19.53 | jasonwoot | boy I wish I hadn't removed the comments from sip.conf.... is the context the same as iax.conf bind? |
| 21:20.00 | [TK]D-Fender | pecanha: 1001 is your SPA? |
| 21:20.49 | Micc | http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation talks about vserver setup. Maybe thats my problem. I'm on a virtual server. |
| 21:21.13 | Micc | but I don't know if I can get them to do the install of the driver on their side. |
| 21:21.17 | pecanha | [TK]D-Fender: yes, I'm connected to 1001 with pap2 |
| 21:21.29 | Micc | can't I use just regular IAX2 channels without the timer? |
| 21:21.39 | [TK]D-Fender | pecanha: allow=ilbc <-- you allwed ilbc when I told you it didn't support it <- |
| 21:21.40 | Micc | It was working fine before. |
| 21:21.48 | [TK]D-Fender | pecanha: disallow=all , allow=ulaw |
| 21:22.11 | [TK]D-Fender | Micc: Indeed zaptel virtualization = trouble |
| 21:22.26 | pecanha | [TK]D-Fender: but If I switch to softphone I'll need to change again, isn't there a way to allow both or I need to create a different extension? |
| 21:22.39 | *** join/#asterisk mattx86 (n=matt@static2073.uctnwd.ken-tennwireless.com) |
| 21:22.58 | marcrosoft | [TK]D-Fender: how did you get to know so much about asterisk? |
| 21:23.14 | [TK]D-Fender | pecanha: I jsut told you what condition is has to be in to work. Configure whatever you want accordingly |
| 21:23.15 | pecanha | marcrosoft: hehe good question |
| 21:23.44 | [TK]D-Fender | marcrosoft: I've been using it for many years now, and I am a private consultant as well |
| 21:23.58 | [TK]D-Fender | marcrosoft: Like everything else, jsut takes some understanding time & dedication. |
| 21:24.06 | marcrosoft | [TK]D-Fender: true |
| 21:24.52 | *** part/#asterisk Chesther (n=cam2@cam2-win.cit.cornell.edu) |
| 21:26.26 | pecanha | [TK]D-Fender: on trunk I can use ilbc,alaw,ulaw... and only on device I put ulaw, it will work? |
| 21:26.57 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.5) |
| 21:27.08 | [TK]D-Fender | pecanha: yes, but you should only use 1 codec per device period. |
| 21:27.14 | Micc | aha, I don't need the timer |
| 21:27.22 | Micc | I had a 7 on the first line of my iax.conf |
| 21:27.27 | Micc | so it wasn't loading. |
| 21:28.16 | meuserj | Ok.. still can't figure it out. When using the IMAP voicemail storage backend, it still sends an e-mail via sendmail. How do I turn that off so that the user doesn't get multiple copies? |
| 21:28.39 | marcrosoft | [TK]D-Fender: would you recommend going the voip route? |
| 21:28.47 | marcrosoft | [TK]D-Fender: is it reliable enough |
| 21:28.54 | [TK]D-Fender | marcrosoft: depends on your precise needs |
| 21:29.16 | marcrosoft | 4-5 people with 2 lines with 3 numbers that roll over |
| 21:29.29 | marcrosoft | 1 is an 800 number |
| 21:31.03 | [TK]D-Fender | marcrosoft: You'd have to compare ITSP packages + ISP costs to the cost of physical lines |
| 21:31.11 | mattx86 | hey guys, I'm trying to setup an IAX2 link between two * boxes and it's only working in one direction; that is, users on the box with the working link can dial us, but we cannot dial them. iax2 show peers shows OK on their box, but Unreachable on ours. here's the iax.conf snippet that's used on both boxes: http://pastebin.com/d41a8e908 |
| 21:31.31 | marcrosoft | [TK]D-Fender: ITSP packages? |
| 21:31.38 | [TK]D-Fender | ~itsp |
| 21:31.39 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:31.39 | pecanha | [TK]D-Fender: I changed to ulaw, but didn't work |
| 21:31.47 | [TK]D-Fender | pecanha: And you aren't showing anything. |
| 21:31.54 | marcrosoft | [TK]D-Fender: ISP costs would be built in as we need that no matter what |
| 21:32.59 | marcrosoft | [TK]D-Fender: phone bills are about $350 a month |
| 21:33.25 | pecanha | sry |
| 21:33.26 | [TK]D-Fender | marcrosoft: Look in serious detail. |
| 21:34.31 | pecanha | http://pastebin.ca/1275320 |
| 21:36.11 | marcrosoft | [TK]D-Fender: well what are the costs involved with having a service that holds the number and transfers to voip |
| 21:36.39 | *** part/#asterisk gpowers (n=glenn@adsl-99-142-75-162.dsl.emhril.sbcglobal.net) |
| 21:36.40 | [TK]D-Fender | ~itsplist-us |
| 21:36.41 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 21:36.43 | *** join/#asterisk linuxviewer (n=nick@ip70-190-194-221.ph.ph.cox.net) |
| 21:37.30 | [TK]D-Fender | pecanha: SIP/2.0 603 Declined |
| 21:39.11 | [TK]D-Fender | pecanha: your DIALPLAN is refusing you now. |
| 21:39.24 | pecanha | [TK]D-Fender: hmmm |
| 21:40.18 | marcrosoft | [TK]D-Fender: what are "channels |
| 21:41.09 | linuxviewer | Currently when someone dials an extension and there isnt a phone or softphone attached and active to that extension, then it goes to an automated "computer sounding" voice saying to leave a message. Is there anyway to get it so that it plays the "unavailable" message? |
| 21:41.27 | [TK]D-Fender | macrDepends. |
| 21:42.04 | [TK]D-Fender | linuxthis is YOUR dialplan. go change how you call Voicemail. |
| 21:42.34 | pecanha | [TK]D-Fender: thanks, I'll stop for today |
| 21:42.52 | pecanha | my brain is melt hehe |
| 21:43.23 | pecanha | bye all |
| 21:44.39 | jaytee | quittin time, be back later |
| 21:44.57 | linuxviewer | what would happen in this situation: i have an extension, example 500, and i have two users logged into extension 500 (softphone and hardphone)... will both ring when someone dials that extension? |
| 21:46.21 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 21:46.29 | Micc | Why would my ssh session freeze after I don't use it for some time. |
| 21:47.35 | Micc | then I hit some keys and type some things then after a few minutes it shows everything and seems normal again. |
| 21:47.42 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:49.10 | Katty | ponders |
| 21:49.14 | Katty | so if i change a sip password in sip.conf |
| 21:49.17 | Katty | sip reload |
| 21:49.23 | Katty | then change the polycom phone's password on the lines section |
| 21:49.31 | Katty | and it spew incorrect password all over the CLI |
| 21:49.33 | *** join/#asterisk mark_csi (n=mark@host86-131-116-42.range86-131.btcentralplus.com) |
| 21:49.36 | Katty | what would i check first :/ |
| 21:49.36 | [TK]D-Fender | linuxviewer: No |
| 21:49.56 | [TK]D-Fender | linuxviewer: If 2 devices are set to register to a given account then the last one through will get the call |
| 21:51.33 | mark_csi | hi everyone - I've changed the latency of my tdm800 card in /etc/modprobe.d/dahdi file, but after I've rebooted it's ignored them. Any ideas? |
| 21:53.10 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 21:53.19 | marcrosoft | [TK]D-Fender: if you wanted 2 phones to ring what you would you do? |
| 21:55.50 | mark_csi | marcrosoft: why don't you just create a ring group? |
| 21:57.41 | [TK]D-Fender | marcrosoft: "core show application dial" <- and set up 2 phones |
| 21:57.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:00.13 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:00.13 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 22:00.55 | [TK]D-Fender | blitzrage: I DON'T WANT TO KNOW YOUR NAME! |
| 22:01.12 | blitzrage | I JUST WANT... |
| 22:01.14 | [TK]D-Fender | blitzrage: ! ! ! |
| 22:02.17 | marcrosoft | [TK]D-Fender & mark_csi i see.. |
| 22:02.32 | marcrosoft | do you still pay taxes with voip ITSP? |
| 22:02.36 | Katty | wibbles. |
| 22:02.59 | Katty | i dun understand. |
| 22:03.06 | Katty | asterisk /shows/ right secret. |
| 22:03.11 | Katty | phone programmed with right password. |
| 22:03.14 | Katty | BUT NO WORKITH |
| 22:03.18 | Katty | cries |
| 22:03.48 | Carlos_PHX | Katty: Could be many things, did you try no password? |
| 22:03.55 | Carlos_PHX | Specific error? |
| 22:04.37 | Katty | i tried a random password with uppercase lowercase number and password, a lowercase only password, 3 digits, and 4 digits. |
| 22:04.45 | Katty | each time sip show users showed the correct password. |
| 22:04.53 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 22:05.01 | Katty | http://pastebin.ca/1275357 <- keeps spewing that |
| 22:05.03 | [TK]D-Fender | Katty: "sip show users" doesn't SHOW the password |
| 22:05.06 | Katty | every phone in the building does the same thing |
| 22:05.14 | Carlos_PHX | It does for me. |
| 22:05.17 | Katty | [TK]D-Fender: secret column? |
| 22:05.34 | [TK]D-Fender | News to me.. |
| 22:05.42 | Katty | [TK]D-Fender: between the username and accountcode column |
| 22:05.43 | Carlos_PHX | I'd show you, but..... |
| 22:05.58 | Carlos_PHX | Username Secret Accountcode Def.Context ACL NAT |
| 22:06.00 | Katty | would MAC-phone.cfg have anything to do with this? |
| 22:06.18 | Carlos_PHX | What's the error? |
| 22:06.23 | Katty | see pastebin above. |
| 22:07.34 | Katty | polycom 330s, btw |
| 22:07.57 | Carlos_PHX | Try a sip debug and see what they actually say. Also have you tried configuring a phone using the web UI? That way you eliminate central config problems. |
| 22:08.10 | Katty | that's the way i do it, through the ip |
| 22:08.19 | Katty | confirming they reboot themselves |
| 22:08.25 | Katty | i'll do a sip debug, see what happens |
| 22:08.35 | Carlos_PHX | You mentioned a phone cfg file before. |
| 22:09.04 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 22:10.10 | Katty | yeah. |
| 22:10.18 | Katty | but i moved them out of the directory temporarily just to see what would happen |
| 22:10.22 | Katty | out of FTP |
| 22:10.24 | Katty | didn't seem to help |
| 22:10.24 | marcrosoft | can you setup asterisk so a skype caller can call it and it rings on a ip phone? |
| 22:12.10 | Katty | okay. i purposely made a bad password on 130 |
| 22:12.29 | Katty | does the debug show the actual password its using? |
| 22:14.22 | Carlos_PHX | I thought it did, but don't truly recall. |
| 22:14.57 | Carlos_PHX | Are you setting the account and auth name both? |
| 22:15.18 | Carlos_PHX | Try a softphone to test the account? |
| 22:15.31 | *** join/#asterisk metfan2007 (n=jc@fw.grupositel.com.mx) |
| 22:16.00 | metfan2007 | Hi all, I don't understand what does "Call failed to go through, reason (3) " means, can you helpme? is that the ring timeout expired? |
| 22:16.16 | metfan2007 | "Call failed to go through, reason (3) Remote end Ringing" |
| 22:18.32 | *** join/#asterisk jer_ (n=jer@unaffiliated/jer) |
| 22:19.24 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:21.35 | mark_csi | marcrosoft: skype and digium are working on this right now, a beta is due for release shortly. |
| 22:21.56 | marcrosoft | mark_csi: sweet |
| 22:22.16 | Katty | Carlos_PHX: i found the problem |
| 22:22.23 | Katty | Carlos_PHX: we don't have any MAC-phone.cfg files here |
| 22:22.29 | Katty | Carlos_PHX: we all use just regular old sip.cfg |
| 22:22.39 | Katty | Carlos_PHX: there, they all have MAC-phone.cfg files |
| 22:22.46 | Katty | Carlos_PHX: with the old password in there |
| 22:23.12 | Katty | Carlos_PHX: do you know if a polycom with automatically write one of those cfg files before reboot |
| 22:23.25 | *** part/#asterisk mark_csi (n=mark@host86-131-116-42.range86-131.btcentralplus.com) |
| 22:27.40 | JonOnt | hey guys, im using FreePBX, was just wondering what section I would deal with how many rings untill a call goes to voice mail after i transfer it to another extention, or rather, what happens when the extention is busy |
| 22:28.28 | [TK]D-Fender | JonOnt: go ask in #freePBX . GUI config is not supported here |
| 22:31.05 | JonOnt | [TK]D-Fender, thank you Fen, you da man |
| 22:32.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:34.28 | Katty | considers renaming to moo.cfg for a wittle testypoo |
| 22:35.29 | sprite-- | I created a user asterisk with /var/run/asterisk as path and /sbin/nologin as their shell. I have astrundir => /var/run/asterisk yet my asterisk.ctl and asterisk.pid is created in /var/run not /var/run/asterisk what am I doing wrong? |
| 22:36.03 | *** join/#asterisk ecrist (n=ecrist@chunk.secure-computing.net) |
| 22:36.16 | *** part/#asterisk ecrist (n=ecrist@chunk.secure-computing.net) |
| 22:41.03 | *** join/#asterisk bkruse (n=bkruse@nat/digium/x-110ea0b5ef125bbf) |
| 22:41.03 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 22:41.50 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 22:44.52 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:53.25 | marcrosoft | does asterisk use sendmail to send out voicemail? |
| 22:55.48 | marcrosoft | i got the voice mail to work.. but no email |
| 22:57.14 | JonOnt | Hey guys, how do I set an extention so that if the person doesnt pick up the call, instead of going to voicemail, the call returns to the main que?. |
| 22:59.08 | jblack | Have two dial lines. The first dials the target. The second dials the queue. |
| 22:59.28 | jblack | possibly replace the second dial with a queue specific operation |
| 22:59.50 | *** join/#asterisk [netman] (n=netman@166.Red-88-23-82.staticIP.rima-tde.net) |
| 23:01.37 | [TK]D-Fender | marcrosoft: Yes, and you have a few settings to do in voicemail.conf for this |
| 23:02.03 | [TK]D-Fender | JonOnt: Wrong channel |
| 23:04.28 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-183.phlapa.east.verizon.net) |
| 23:04.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:04.53 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 23:07.00 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:08.01 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 23:08.44 | harry_v | I was wondering why my dial plan is not excepting callerid enabled for calls where the called party requires it. Example, I would append *82 to NXXNXXXXXXX,1 but the call will not go out. Not sure if there should be a delay set aside before the other digits are dialed or what might be the case. |
| 23:09.35 | harry_v | exten => _*82NXXNXXXXXX,1,Dial(dahdi/1/${EXTEN}) |
| 23:09.35 | harry_v | that should work |
| 23:11.16 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:11.36 | harry_v | hi jaytee |
| 23:12.18 | jaytee | hi |
| 23:13.25 | harry_v | tell me something, appending *82 to a nomral dialout of NXXNXXXXXXX should enable caller id right? |
| 23:13.38 | sprite-- | jaytee: I have astrundir set to /var/run/asterisk but it tries to create my pid file in /var/run/ instead. any idea? |
| 23:14.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:14.26 | harry_v | Because I have that in my dialplan and it will not send it as output with the 11 digits on cli. |
| 23:15.33 | [TK]D-Fender | harry_v: that is PREPENDING, and you may need a pause, etc |
| 23:15.45 | [TK]D-Fender | harry_v: And assuming that it is an analog channel |
| 23:15.51 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-76-209.tx.res.rr.com) |
| 23:17.13 | harry_v | That was what I was thinking. |
| 23:17.18 | sdaniels | can someone point me to an example of how to do something with a call based on source #, is there something like if (ani == 6305555555){ do this stuff } else { do this stuff } ? |
| 23:17.23 | harry_v | So how to introduce the pause |
| 23:17.44 | jaytee | harry_v, [TK]D-Fender is correct as usual. You need to prepend the *82 and that assumes that your default CALLERID state is set to block. |
| 23:18.23 | [TK]D-Fender | sdaniels: "core show application gotoif" |
| 23:18.29 | sdaniels | th |
| 23:18.31 | sdaniels | thx |
| 23:18.33 | [TK]D-Fender | sdaniels: "core show function callerid" |
| 23:18.45 | [TK]D-Fender | sdaniels: and go read up on "Asterisk Expressions" on the WIKi & in the BOOK |
| 23:19.11 | sdaniels | got the book, someone stands to make alot of money by writing a good one. |
| 23:19.19 | [TK]D-Fender | ~book |
| 23:19.19 | jbot | book is probably probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:19.44 | [TK]D-Fender | sdaniels: Guess your statement was intentionally double-edged. |
| 23:20.01 | sdaniels | [TK]D-Fender: aye |
| 23:20.09 | jaytee | sdaniels, someone stands to lose alot of time and not make much money writing a "good" one. The one we have is pretty decent if people actually took the time to read it. |
| 23:20.46 | jaytee | now a Harlequin romance novel, in spite of how insipidly stupid and predictable, would be a real money maker. |
| 23:20.53 | sdaniels | funny how different channels have different personalities |
| 23:22.16 | sprite-- | Nevermind figured out my problem. |
| 23:22.27 | *** join/#asterisk qdk (n=qdk@212.27.24.104.bredband.3.dk) |
| 23:22.51 | jaytee | sprite, sorry....got caught up in something else. what'd it turn out to be? |
| 23:24.22 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 23:24.26 | sprite-- | Didn't edit out the (!) in asterisk.conf |
| 23:24.47 | [TK]D-Fender | sprite--: You meant he one with the giat text warning above it? |
| 23:25.02 | sprite-- | Got Asterisk 1.6.0.2 up and running on my Gentoo box though, think I'm just going to scrap my AsteriskNOW upgrade and put Gentoo on that as well. |
| 23:26.04 | [TK]D-Fender | sprite--: Cool... can I have your copy of chan_fluxcapacitor.so wihle you're at it? |
| 23:26.42 | sprite-- | [TK]D-Fender: Thought this was supposed to be a support channel, do you really need that holier than thou attitude just because someone is new at something? |
| 23:27.25 | [TK]D-Fender | sprite--: for the "warning" comment, take it as the comedic jab that it was... |
| 23:27.52 | [TK]D-Fender | sprite--: Sorry if I don't put a smiley on everything that should be taken lightly(er) |
| 23:28.07 | sprite-- | Hah no problem, I'm probably just tired :) |
| 23:28.43 | [TK]D-Fender | sprite--: You've been at things for quite a while, I'm sure you are. |
| 23:33.14 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:33.16 | *** join/#asterisk Segnale007 (n=Pietro@host71-242-dynamic.9-79-r.retail.telecomitalia.it) |
| 23:36.43 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 23:37.37 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 23:37.50 | sdaniels | GotoIf($["${CALLERID(num):0:3}" = "877"]?1000) <----- does the 0:3 represent from the beginning to the 3rd character of num? |
| 23:38.00 | *** part/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 23:38.33 | bkruse | sdaniels: yes |
| 23:38.45 | bkruse | if 877 number, go to 1000 |
| 23:38.53 | sdaniels | bkruse: so what would 1:3 do? |
| 23:39.01 | sdaniels | im trying to understand the 0 |
| 23:39.02 | bkruse | 1st character to 3rd character |
| 23:39.15 | bkruse | Just go |
| 23:39.35 | bkruse | NoOp(${CALLERID(num):0:3}) |
| 23:39.37 | bkruse | NoOp(${CALLERID(num):1:3}) |
| 23:39.57 | sdaniels | so thats the same thing right? |
| 23:41.04 | sdaniels | seems to me that 0:3 is 4 characters |
| 23:41.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 23:42.27 | sdaniels | unless 0 just represents the begining |
| 23:42.30 | Corydon76-dig | Second digit is a length, not an offset |
| 23:42.44 | sdaniels | ahhh |
| 23:43.58 | jaytee | of course I had to actually read page 140 of the book and look at the examples before I understood how that worked. |
| 23:43.58 | sdaniels | so from 0 (the first character) count 3 characters |
| 23:44.08 | sdaniels | so 1:3 wold be from the 2nd character count 3 characters |
| 23:44.14 | sdaniels | got it thanks. |
| 23:44.15 | Corydon76-dig | Correct |
| 23:44.19 | jaytee | sheer genius! |
| 23:44.26 | sdaniels | jaytee: you gettin royalties off them or somethin bro? |
| 23:45.46 | Corydon76-dig | and in 1.4, you can use a negative offset, too |
| 23:46.16 | jaytee | for every book I pimp I get 10 S&H Green Stamps. I'm saving up for the Winnebago. Only need 247,582,234,444,330 more to go! |
| 23:46.46 | sdaniels | so if 1234 3:-3 would that be 432 or 234? |
| 23:46.54 | sdaniels | jaytee: lol |
| 23:47.46 | Corydon76-dig | Negative offset, not negative length |
| 23:48.02 | Corydon76-dig | although, you can do negative length, too |
| 23:48.06 | jaytee | meaning the minus can only be in the first position |
| 23:48.12 | jaytee | you can? |
| 23:48.26 | Corydon76-dig | negative offset means start n from the right end |
| 23:48.44 | Corydon76-dig | negative length means end n before the end of the string |
| 23:49.16 | sdaniels | oh thats cool so if you have some long ass string you dont have to count from the left |
| 23:49.23 | Corydon76-dig | so 1234:3:-3 would be blank |
| 23:49.28 | *** join/#asterisk blinky42 (n=sbrown@67.200.59.44) |
| 23:49.46 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 23:49.47 | Corydon76-dig | It would only give you something if the string was at least 7 characters long |
| 23:50.09 | jaytee | so with 12345678 as the string ${EXTEN:1:-4} would return 234? |
| 23:50.19 | sdaniels | i dont get the negative offset then... how is that usefull? |
| 23:50.25 | Corydon76-dig | jaytee: correct |
| 23:50.37 | Corydon76-dig | sdaniels: 1234:-1 gives 4 |
| 23:50.44 | sprite-- | jaytee: Sorry to keep bothering you. Trying to get Asterisk 1.6 to work on my new box, when I call in from my cell I get [Dec 3 18:49:16] NOTICE[17598]: chan_sip.c:16983 handle_request_invite: Call from '' to extension '17772784063' rejected because extension not found. |
| 23:50.46 | jaytee | sdaniels, it means start from the "right" end instead of the left |
| 23:50.49 | *** join/#asterisk WindBack (i=jorge@201-213-250-41.net.prima.net.ar) |
| 23:51.04 | sdaniels | oh ok duh you said that... i get it. |
| 23:51.22 | jaytee | sprite, it means that * can't find an extension in the context you have for your inbound route for that call |
| 23:51.58 | Corydon76-dig | Honestly, I wasn't sure the value of negative length, other than I wanted it to do something nice if somebody tried that |
| 23:52.07 | sprite-- | the only extension I have defined in there is s.... |
| 23:52.14 | jaytee | the book doesn't show a negative length example though. They should put that in the 3rd edition |
| 23:52.39 | Corydon76-dig | jaytee: probably because it's somewhat confusing |
| 23:52.53 | sprite-- | jaytee: Shouldn't call from '' not be blank? |
| 23:52.54 | tzanger | negative length? |
| 23:52.55 | Corydon76-dig | Neither negative worked in 1.2 |
| 23:52.58 | jaytee | Corydon76-dig, yeah it took me a few to wrap my head around that |
| 23:53.03 | sdaniels | honestly the book is good to get started, but I learn much faster from asking real people that know more than me questions. |
| 23:53.17 | jaytee | sprite, the " is an empty callerid field |
| 23:53.28 | Corydon76-dig | tzanger: negatives in the ${::} syntax |
| 23:53.56 | jaytee | meaning either you're using an analog line that doesn't pass CID info or the caller has CID blocked or sumthin else. |
| 23:54.20 | sprite-- | Well I was calling from my cell and it was passing CID earlier today.... |
| 23:54.54 | jaytee | sprite, if it's default is to pass it then after any call where you've blocked it with *67 it should reset |
| 23:55.22 | sdaniels | anyone know of a sip provider that sends rdnis with the callerid? |
| 23:55.46 | jaytee | sprite, how is the call coming in? analog? SIP? PRI? |
| 23:56.30 | sprite-- | SIP |
| 23:57.51 | jaytee | and in the context you have your sip provider pointed to in extensions.conf you need to set a pattern mask or exact extension match. s is only good for analog lines for FXO ports or for macros. |
| 23:59.00 | sprite-- | Ahhh ok. |
| 23:59.33 | jaytee | sprite, so change the s in that context to 17772784063, do a dialplan reload and retest your call |