| 20:52.37 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 20:52.37 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 20:53.48 | ZcCOP | right.. i guess im failing to see where that gets pumped in.. Server B is running realtime extensions, but since the PRI server passed the call using a macro, server B sees the extension as something arbitrary, and not the actual number |
| 20:55.05 | ZcCOP | and theres multiple numbers pointing to this end user, so I was hoping to do something that would enable me to easily point many DID's to a single peer |
| 20:56.56 | ZcCOP | is there a way in extensions.conf to Dial() a dundi peer directly?? like Dial(DUNDI/5551212@sippeer) ?? |
| 20:57.44 | hardwire | does 1.6 have better log timestamps and unique id correlation? |
| 21:03.25 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 21:03.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 21:12.57 | *** join/#asterisk dwxreaper (n=quinn@cvsweb.activehost.com) |
| 21:13.13 | ManxPowerAsteria | ZcCOP: since DUNDI is not a VoIP protocol, the answer is NO! |
| 21:13.15 | dwxreaper | how do I turn off debug output to the asterisk cli? |
| 21:13.36 | dwxreaper | and another question :), is there an snmp oid to monitor queue length for call queues? |
| 21:13.40 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.195) |
| 21:13.49 | stabler | dwxreaper: are you refering to sip debug? |
| 21:14.07 | dwxreaper | stabler: I get a bunch of messages all the time at the asterisk cli, i'm not sure what they are, they look like sip debugs |
| 21:14.16 | stabler | sip set debug off |
| 21:14.22 | dwxreaper | stabler: i'd like to turn all the output to the cli off, so I just connect and get the prompt |
| 21:15.07 | dwxreaper | stabler: does that turn off all possible debugging, anyway I can log to a file and just not standard out or whatever |
| 21:15.08 | stabler | dwxreaper: alot of those outputs are really useful.. i dont suggest turning "normal" outputs off |
| 21:15.22 | dwxreaper | stabler: i'm going to run an expect script so I want them all off |
| 21:15.28 | dwxreaper | I'd like to log them to a file or just turn them all off |
| 21:15.31 | ManxPowerAsteria | dwxreaper: I think you think "debug" means something other than what it means. |
| 21:15.48 | dwxreaper | ManxPowerAsteria: I just want them off |
| 21:15.53 | ManxPowerAsteria | maybe you means "don't give me CLI output" |
| 21:16.14 | dwxreaper | I want the output off so it's like a normal shell.. |
| 21:16.22 | ManxPowerAsteria | you turn debug messages off by "set debug off" or "core set debug off" but you will still get warning messages, error messages, notice messages, and application output. |
| 21:16.36 | ManxPowerAsteria | dwxreaper: edit /etc/asterisk/logger.conf and comment out the console => line. |
| 21:16.42 | ManxPowerAsteria | or set debug off and set verbose 0 |
| 21:17.00 | dwxreaper | ManxPowerAsteria: from the cli, just type "set debug off" and "set verbose 0" ? |
| 21:17.42 | *** join/#asterisk xchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 21:17.51 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 21:18.10 | dwxreaper | ManxPowerAsteria: that's just related to the output, nothing with the actual calls right? |
| 21:18.25 | ManxPowerAsteria | dwxreaper: do you see messages with the word DEBUG at the start? |
| 21:18.33 | dwxreaper | ManxPowerAsteria: no |
| 21:18.46 | dwxreaper | ManxPowerAsteria: so I have to edit out console => line? |
| 21:18.47 | ManxPowerAsteria | then you don't have debug messages enabled and trying to set it to off or 0 will do nothing at all. |
| 21:19.29 | ManxPowerAsteria | dwxreaper: the only way to get no messages on the console that I know of is to remove the console => line or maybe just remove all the message levels from that line. |
| 21:19.50 | ManxPowerAsteria | dwxreaper: the term you are looking for is "CONSOLE MESSAGES" |
| 21:19.58 | ManxPowerAsteria | using the term debug messages just wasted everyone's time. |
| 21:20.07 | dwxreaper | ManxPowerAsteria: whatever |
| 21:21.34 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 21:23.14 | *** join/#asterisk drjones1 (n=drjones@75.110.209.55) |
| 21:23.19 | drjones1 | Hey guys |
| 21:23.34 | drjones1 | anyone wana buy or know where i can sell 16 cisco 7960s w/ sip firmware? |
| 21:24.06 | ManxPowerAsteria | asterisk-biz mailinglist |
| 21:24.29 | MaliutaLap | drjones1: ebay |
| 21:24.47 | drjones1 | mannn ebay is such a hassle |
| 21:26.25 | j_kroon | hi all, i'm having some issues compiling app_fax.c in asterisk 1.6.0.3, it's complaining about unknown size of fax_state_t. |
| 21:26.27 | j_kroon | any ideas? |
| 21:27.46 | MaliutaLap | drjones1: you can pay the postage and give me $100 to take them off your hands then :P |
| 21:27.52 | dwxreaper | is there an OID for queue length? |
| 21:28.16 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 21:28.27 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:28.37 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.10.119) |
| 21:30.13 | ManxPowerAsteria | j_kroon: I assume this is the fax app that is included in 1.6? If so then chances you have the wrong version of SpanDSP |
| 21:33.46 | stabler | drjones1: ill pay postage and givey you $100 |
| 21:33.58 | drjones1 | per phone/ |
| 21:33.59 | drjones1 | ? |
| 21:33.59 | drjones1 | :) |
| 21:34.04 | stabler | just beat MaliutaLap's bid! |
| 21:34.11 | j_kroon | ManxPowerAsteria, ok, which versions does work? I think I've got 0.0.6 installed. |
| 21:34.24 | ManxPowerAsteria | j_kroon: whatever is recommended in the docs |
| 21:34.41 | MaliutaLap | drjones1: sure, you can pay me $100/phone |
| 21:34.41 | j_kroon | goes off to find the docs. |
| 21:34.58 | MaliutaLap | drjones1: are we talking $US or $AU?? |
| 21:35.00 | stabler | drjones1: im not a big fan of cisco phones w/sip firmware |
| 21:35.40 | Deeewayne | j_kroon, if you can work with the 1.6.0 branch instead of an older tag, all versions of spandsp will work |
| 21:35.59 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 21:36.09 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 21:36.34 | j_kroon | Deeewayne, app_fax.c: In function 'transmit_audio': ... app_fax.c:312: error: storage size of 'fax' isn't known |
| 21:36.37 | j_kroon | it's type is fax_type_t. |
| 21:36.45 | j_kroon | fax_state_t sorry. |
| 21:36.57 | nullable_type | Hey guys, Is there a way i can call Originate through http instread of AGI from a remote PC? |
| 21:36.58 | stabler | drjones1: ebay is probably your best option |
| 21:37.16 | drjones1 | think I should try to sell them all in one lot? |
| 21:37.21 | drjones1 | or indiviudually |
| 21:37.23 | drjones1 | and does ebay charge you |
| 21:37.27 | drjones1 | if it doesn't sell? |
| 21:37.28 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:37.44 | ManxPowerAsteria | yes they charge you no matter what |
| 21:37.46 | stabler | drjones1: yes ebay charges reguardless |
| 21:38.15 | Deeewayne | j_kroon, the spandsp api has been changing across 0.0.6preX releases. If you are going to use Asterisk 1.6.0.3, you will have to use a specific version of spandsp |
| 21:38.18 | stabler | sell them one at a time but allow the winner to choose how many they want to buy |
| 21:38.35 | stabler | if that makes sense |
| 21:39.09 | stabler | i dont know the technical ebay name for that kind of auction |
| 21:39.22 | j_kroon | Deeewayne, thanks. I'm guessing 0.0.5 something? |
| 21:39.30 | ManxPowerAsteria | you can select it when you create the auction |
| 21:39.45 | stabler | ManxPowerAsteria: yes |
| 21:40.04 | stabler | ManxPowerAsteria: i just dont remember what they call it |
| 21:40.45 | Deeewayne | j_kroon, maybe. |
| 21:42.05 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 21:42.13 | nullable_type | Hey guys, Is there a way i can call Originate through http instread of AGI from a remote PC? |
| 21:42.41 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 21:45.34 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:45.34 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:46.01 | nullable_type | Asterisk Manager Access via HTTP, Have you guys used it much as a replacement of AGI? |
| 21:46.05 | lmadsen | anyone setup IMAP with Asterisk before? I'm configuring asterisk for dovecot, but not sure where to create the folder that is configured in imapfolder= within voicemail.conf |
| 21:46.06 | bmoraca | nullable_type: sure...use a PHP (or whatever other server-side scripting you want) to drop a call file in the spooler directory |
| 21:46.23 | lmadsen | huh? AMI != AGI |
| 21:46.37 | lmadsen | so I'm not sure how you use AMI to replace AGI... |
| 21:47.19 | nullable_type | ya sorry, I meant just things like Originate Call, instead of using TCP/IP Sockets, i would like to use HTTP |
| 21:47.23 | nullable_type | I guess AMI is the solution? |
| 21:47.25 | nullable_type | thanks |
| 21:50.47 | nullable_type | bomoraca >> Did you mean i can use HTTP or do i need a PHP Script in the asterisk server to do that? |
| 21:53.28 | codefreeze-lap | Uh, one of my boys brings up an interesting question: what do you do with text messages? For instance, via chan_mobile, if an SMS came in, and your phone could handle passing it to the chan_mobile driver, what do you do with it then? |
| 21:58.44 | stabler | codefreeze-lap: asterisk can handle the text messages but im not sure how to do it |
| 21:59.37 | stabler | codefreeze-lap: i bet some quick googleing would reveal an answer |
| 22:00.22 | codefreeze-lap | stabler: true... just wondering if/how others were doing it here... apparently, from all the blank stares, not may are... |
| 22:00.37 | codefreeze-lap | s/not may/not many/ |
| 22:00.45 | stabler | codefreeze-lap: i had plans to play with chan_mobile but never got around to it |
| 22:01.23 | codefreeze-lap | stabler: in the case of chan_mobile, it might only handle sending text out; haven't read it thru yet on that subject. |
| 22:01.41 | stabler | codefreeze-lap: im positive that it will revieve |
| 22:02.02 | stabler | *receive |
| 22:03.09 | stabler | codefreeze-lap: if i remember correctly i believe most poeple just have it output to the cli |
| 22:03.32 | stabler | codefreeze-lap: but im sure there are alot of cool things you could do with it ;) |
| 22:05.06 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:12.45 | drjones1 | i also have some diguim cards for sale |
| 22:12.47 | drjones1 | and plantronic headsets |
| 22:13.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:14.18 | j_kroon | tzafrir_laptop, thanks for the help last night. really appreciated. |
| 22:15.17 | *** part/#asterisk mastershake64 (n=AJ@c-71-58-112-242.hsd1.pa.comcast.net) |
| 22:20.26 | codefreeze-lap | stabler: yeah, you are right. chan_mobile can receive SMS messages via bluetooth; It fires up a new fake mobile channel; sets the SMSSRC and SMSTXT vars to the incoming data, and sets the exten to "sms", so it will try to execute the dialplan at the normal chan_mobile context, with sms extension. I guess there you could read out those vars, and maybe jingle or gtalk or email it to yourself or something. |
| 22:21.24 | j_kroon | o.O |
| 22:21.27 | j_kroon | shweet! |
| 22:23.27 | voxter | lmadsen: ping! |
| 22:25.30 | lmadsen | voxter: pong! |
| 22:27.04 | voxter | lmadsen: you mainly work w/ polycom handsets, right? |
| 22:27.22 | minotaur01 | has anyone setup asterisk sms on at pri connection? |
| 22:27.41 | voxter | minotaur01: how would that work exactly? :) |
| 22:27.43 | ManxPowerAsteria | minotaur01: You would have to find some european people for that. |
| 22:28.00 | voxter | Oh, right. europeans and their ubiquitous SMS. |
| 22:28.36 | ManxPowerAsteria | app_sms uses a fsk burst to talk to a SMS Service Center |
| 22:28.51 | ManxPowerAsteria | (or SMS device with PSTN support) |
| 22:29.04 | lmadsen | voxter: kinda ya :) |
| 22:30.05 | voxter | lmadsen: two questions then. #1, what do you use for headsets (corded and cordless) and #2 what other phones have you started using? :) |
| 22:30.42 | nullable_type | Hey guys i was using HTTP For AMI. It seems i have to login first to do an originate command. Can't i just pass login info along with originate command, to avoid the Cookie Session? |
| 22:30.52 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.100.102) |
| 22:30.59 | DelphiWorld | hello my friends |
| 22:31.17 | DelphiWorld | please any VoIp community or company give me a free IAX acount ? |
| 22:31.36 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 22:33.07 | minotaur01 | voxter: i don't know how it would work thats why im asking i read a article saying that you can do it but can't find any info on how to make it work |
| 22:33.21 | jeremy_g | DelphiWorld:we only have sip account and a policy of giving them. |
| 22:33.27 | jeremy_g | accounts |
| 22:33.30 | voxter | minotaur01: all (most?) pri providers in north america do not provide SMS services via PRI |
| 22:33.53 | lmadsen | voxter: honestly, that's not really my problem to deal with. I don't really do deployments anymore |
| 22:33.59 | lmadsen | and I've never had to buy a headset |
| 22:34.19 | voxter | lmadsen: ah, gotcha. I thought you still did deployments. Must be nice not to deal with the end user :) |
| 22:34.31 | ManxPowerAsteria | minotaur01: generally you can only use SMS over PSTN to connect to european service providers. |
| 22:34.33 | wtca | having a problem with playing greetings from IMAP. It would seem that greetings get stored fine but they can not be retrieved.. Running latest svn version. Any thoughts ? |
| 22:34.35 | minotaur01 | voxter: okay. that answers my question. |
| 22:34.48 | nullable_type | Is anone here can help me with http AMI |
| 22:35.01 | lmadsen | voxter: haven't really done deployments in quite some time... and when I did, it was really for larger scale stuff where I wasn't interacting with users really |
| 22:35.25 | minotaur01 | ManxPowerAsteria: i want to send sms to NA providers |
| 22:35.30 | lmadsen | nullable_type: just ask the specific question you'd normally ask |
| 22:35.51 | minotaur01 | ManxPowerAsteria: our customers are in NA |
| 22:35.57 | ManxPowerAsteria | minotaur01: And I want $1 million and a couple of strippers. Neither of us will get what we want. |
| 22:36.01 | jeremy_g | minotaur01:where are you based at |
| 22:36.16 | minotaur01 | ontario |
| 22:36.21 | ManxPowerAsteria | minotaur01: for NA SMS you would want to sign up for a service provider. |
| 22:36.36 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:36.51 | jeremy_g | gives [TK]D-Fender a cold shower |
| 22:37.32 | jeremy_g | minotaur01:whats the problem in setting up sms over pri. i mean whats so special about sending it over a "PRI" |
| 22:37.35 | minotaur01 | ManxPowerAsteria: service provider like a telco or another type?? |
| 22:37.40 | [TK]D-Fender | jeremy_g: Few people are qualified to talk to me about "cold". |
| 22:38.07 | ManxPowerAsteria | minotaur01: if you try to get direct SMS service from a carrier they will laugh at you. There are a zillion SMS service providers out there. |
| 22:38.08 | jeremy_g | [TK]D-Fender:uhhh |
| 22:38.10 | minotaur01 | jeremy_g: i have a PRi connection and i want to send SMS to our customers |
| 22:38.51 | jeremy_g | minotaur01:can you send IP traffic on that PRI |
| 22:39.36 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 22:39.49 | minotaur01 | ManxPowerAsteria: whats my other choice using a cell phone or pstn line |
| 22:39.49 | jeremy_g | minotaur01:I did it using kannel in an IP net |
| 22:39.59 | jeremy_g | long time ago |
| 22:40.08 | ManxPowerAsteria | PRI = PSTN |
| 22:40.34 | minotaur01 | i mean an POTS line sorry |
| 22:41.31 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 22:41.46 | blebleble | i'm using dynamic agents, how can i tell if an agent is logged in or not? not seeing it on the command line |
| 22:41.53 | *** join/#asterisk setunado (n=fabien@setuns.fr.nf) |
| 22:42.28 | [TK]D-Fender | blebleble: Show us what you're looking at |
| 22:42.31 | jeremy_g | blebleble:one way could me to add a NoOp when an agent logs and grep that in the log |
| 22:43.09 | stabler | codefreeze-lap: good to hear |
| 22:43.20 | jeremy_g | blebleble:He is referring to the movie running in the background on your PC |
| 22:44.02 | blebleble | no way on the asterisk cli to tell what users are logged in or not? |
| 22:45.22 | wtca | [Mar 2 14:44:47] ERROR[26950]: app_voicemail.c:1189 imap_retrieve_greeting: Voicemail state not found! |
| 22:45.41 | wtca | From a tcpdump it doesn't look like asterisk tries to connect to get greetings |
| 22:47.50 | jaytee | quittin time for me, bbiab |
| 22:48.26 | jeremy_g | [TK]D-Fender: Mind taking a look at bug#14539. |
| 22:48.46 | [TK]D-Fender | jeremy_g: If you don't make me hunt it down |
| 22:49.48 | jeremy_g | [TK]D-Fender:http://bugs.digium.com/view.php?id=14539 |
| 22:50.14 | blebleble | [TK]D-Fender, bascially i'm using dynamic agents (hot desking or whatever you want to call it) and I have users login and logoutof their phones and each extension is statically in a certain queue (customer service) however I am finding that I am unable to determine whether or not a user is logged in or out of the system. The queues and things are working properly so Im sure its being reported somewhere however not seeing any type of command to list it |
| 22:51.20 | jeremy_g | [TK]D-Fender:ain't hunting fun |
| 22:52.08 | jeremy_g | blebleble:have you searched the voip wiki |
| 22:52.13 | [TK]D-Fender | blebleble: what part of "show me" is proving difficult? |
| 22:52.49 | blebleble | pbx01test*CLI> show me |
| 22:52.54 | blebleble | No such command 'show me' (type 'help show me' for other possible commands) |
| 22:53.10 | [TK]D-Fender | ... |
| 22:53.13 | Qwell | blebleble: You just won IRC. and life. |
| 22:54.41 | [TK]D-Fender | blebleble: http://tinyurl.com/496svm |
| 22:54.41 | [TK]D-Fender | jeremy_g: would be nice to see REAL SIP DEBUG for that call. |
| 22:54.41 | [TK]D-Fender | jeremy_g: trust = 0 |
| 22:59.20 | keith4_ | [TK]D-Fender: i'm having some trouble..... |
| 22:59.20 | keith4_ | *CLI> fix it fix it fix it |
| 22:59.20 | keith4_ | No such command 'fix it fix it fix it' (type 'help fix it' for other possible commands) |
| 22:59.22 | blebleble | [TK]D-Fender: I guess i am confused by 'show me' for there is nothing to show thats the whole issue, I have no way of displaying who is logged in or not, thus why i wasnt getting what you were wanting to look at |
| 23:06.11 | keith4_ | which "users" are "logged in"? |
| 23:06.12 | [TK]D-Fender | [17:41]<blebleble>i'm using dynamic agents, how can i tell if an agent is logged in or not? not seeing it on the command line |
| 23:06.12 | [TK]D-Fender | [17:42]<[TK]D-Fender>blebleble: Show us what you're looking at |
| 23:06.12 | [TK]D-Fender | blebleble: SHOW ME your attempt to look at your agents status <- |
| 23:06.12 | blebleble | show queues; sip show users; i went through basically all the show commands and can not find anything that lists users who are actually logged in or logged off |
| 23:06.12 | [TK]D-Fender | blebleble: PASTEBIN! |
| 23:06.12 | blebleble | http://pastebin.ca/1351412 |
| 23:06.13 | [TK]D-Fender | blebleble: Local/554@from-internal/n (Invalid) has taken no calls yet <- this is not an "AGENT" |
| 23:06.13 | [TK]D-Fender | blebleble: it is randome dialplan |
| 23:06.13 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 23:06.13 | [TK]D-Fender | blebleble: With no preconditions attached to it |
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| 23:08.28 | stabler | =/ |
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| 23:09.20 | Bentley | hi all, I want to block all calls coming in on zap that have a certain CID pattern. I'm trying s/_860983XXXX for the extension, but it doesn't match this. Am I using the "_" properly with this diaplan pattern for cid? |
| 23:09.20 | blebleble | maybe the nomenclature is getting funny here, i have 2 phones setup with device and users, 1 phone 554 is static to that phone, the other phone has users set to AdHoc, where 553 and 552 and login and logout of, i can not tell when 552 is logged in or not |
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| 23:09.22 | [TK]D-Fender | Bentley: I'm not sure you can do patterns with CID detect like that. |
| 23:09.22 | Bentley | D-Fender: hmm, ok - too bad .. these damn telemarketers have banks of numbers |
| 23:09.22 | [TK]D-Fender | blebleble: LOCAL. You are dialing a local channel. this is just dialplan, not an AGENT. it has no concept of "logged in" associated to it. There is no such thing as "status" for it. |
| 23:09.22 | [TK]D-Fender | Bentley: Use a GotoIf... |
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| 23:11.15 | blebleble | [TK]D-Fender maybe thats where I'm getting mixed up, I was seeing "users" as "agents" |
| 23:11.25 | [TK]D-Fender | blebleble: not USERS. these are MEMBERS |
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| 23:15.00 | stabler | [TK]D-Fender: whats the differnce in an agent and member? |
| 23:15.52 | [TK]D-Fender | stabler: An Agent is CHAN_AGENT and has login-ins, and other rules attached and need not have ANYTHING to do with Queues. Queues can have MEMEBERS, some of which can be AGENTS |
| 23:16.51 | stabler | [TK]D-Fender: oh ok.. thanks alot! |
| 23:17.04 | blebleble | [TK]D-Fender: for 552, I'm doing *11 putting in 552 and its password and then the phone then turns into 552, if i call 554 says its comming from 552 etc, this is what im talking about, i have no way of monitoring if that 552 user logged in or not |
| 23:17.44 | [TK]D-Fender | blebleble: what are these magic #'s you're talking about? My dialplan has no such thing |
| 23:18.28 | blebleble | sorry *11 and *12 login and logoff default for freepbx |
| 23:18.45 | blebleble | and here is where i need to leave and go to freepbx |
| 23:18.52 | [TK]D-Fender | FUCK FREEPBX AND THE HORSE IT RODE IN ON! |
| 23:19.01 | [TK]D-Fender | blebleble: CORRECT |
| 23:19.17 | [TK]D-Fender | blebleble: Their junk, not *'s |
| 23:19.20 | rob0 | Accomodations provided by the Horseshoe Road Inn. |
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| 23:20.04 | blebleble | yah i was trying to ask to what i saw as a general generic asterisk question, thus why i came here first, everything works properly i just cant tell if someone is logged in or not and wasnt sure if their was a CLI command i was just missing |
| 23:20.33 | [TK]D-Fender | blebleble: "logging in" is a conceptual illusion as processed by that dialplan. |
| 23:20.42 | [TK]D-Fender | blebleble: Ands something we don't support here |
| 23:20.49 | blebleble | ok, thats my confusion then thanks |
| 23:21.06 | [TK]D-Fender | NEXT@!!@!@ (c) BKW |
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| 23:40.55 | wtca | Can anyone tell me what " get_vm_state_by_mailbox " does and if it fails what does it mean ? |
| 23:41.25 | bmoraca | the extra 4 pin molex on a Sangoma A200 TDM card...is that absolutely required to use the card for FXS ports, or is it just recommended? |
| 23:41.44 | [TK]D-Fender | bmoraca: Required |
| 23:41.52 | bmoraca | that's what i though |
| 23:41.54 | bmoraca | t |
| 23:41.56 | fatnasty1 | can i reload asterisk from outside the asterisk CLI? |
| 23:42.14 | bmoraca | damn HP DL380s don't have internal molex connectors |
| 23:42.27 | drjones1 | okay i got my items listed |
| 23:42.29 | drjones1 | i hope they sell |
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| 23:42.53 | [TK]D-Fender | fatnasty1: Where outside? |
| 23:43.14 | fatnasty1 | [TK]D-Fender: I want to reload it with a python script |
| 23:43.30 | [TK]D-Fender | fatnasty1: asterisk -rx "reload" |
| 23:43.41 | fatnasty1 | [TK]D-Fender: excellent, thanks |
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| 23:48.30 | jaytee | [TK]D-Fender, just cutover my inbound PRI to my * server and the Nortel is now sitting idles with no calls in our out, completely cut off from the rest of world :-) \o/ |
| 23:48.49 | Talkradio | :)) |
| 23:49.39 | beek | jaytee: That's got to make you feel good. |
| 23:50.55 | jaytee | beek, you betcha! specially since the span 1 card on my Nortel went into red alarm about 3pm this afternoon and was dropping calls throughout the day intermittently. It was even skewing the secondary card and forcing the TE212P card in the * box to do a reset. |
| 23:51.26 | beek | jaytee: So it gave it's last gasp just as you were going to throw the switch? It must have known... |
| 23:51.34 | [TK]D-Fender | jaytee: \o/ |
| 23:51.55 | beek | evening [TK]D-Fender |
| 23:52.51 | hardwire | how can you request extension 's' over E&M? |
| 23:52.53 | jaytee | beek, no it's the old Nortel way of doing things. They have their PRI's setup as "loops" and they store lof and bipolar counts and if they exceed a set threshold within a fixed time frame they go into alarm. The maintenance modem has been disconnected for over a week so it finally hit threshold today. |
| 23:53.04 | hardwire | by not sending digits at all.. right? |
| 23:53.11 | hardwire | open line.. wink.. done |
| 23:53.35 | beek | jaytee: I see. So when do you power it down and forklift it? |
| 23:54.38 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 23:54.51 | jaytee | beek, in a few weeks time. going to take it apart slowly. The crossconnect map was horribly inaccurate and out of date when I started at this job 6 years ago and I've only been able to map 1/2 of it out so I'm being careful when I start tearing things out. |
| 23:55.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:55.37 | msetim | Who there is using 1.4.23 with agents + agent record enabled + updatecdr=yes? |
| 23:56.56 | Talkradio | bmoraca on all the dells i used a200's in i had to solder in a molex end and cutoff one of the sata power connectors |
| 23:56.57 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 23:57.32 | Talkradio | red/red both black/black and white/yellow |
| 23:58.04 | beek | jaytee: It will be satisfying when it's done. |
| 23:58.55 | jaytee | beek, yep. it's satisfying now but it'll feel even better when there's a big gaping blank space on the wall where the two cabinets for the Option 11C "were". |
| 23:59.39 | Talkradio | or buy one of these which i couldn't find locally http://store.pchcables.com/sa15pinfetom.html |