| 00:02.24 | rue_work | hmm users cant edit their speed dials from the webpage on the polycom phone |
| 00:02.30 | rue_work | thats stupid |
| 00:03.36 | rue_work | the polycom web interface is stupid, you can even reboot the phone with it, its like they stuck it in just to say that its there |
| 00:04.02 | *** join/#asterisk docelmo (n=vircuser@pool-141-152-199-236.lyn.east.verizon.net) |
| 00:04.44 | *** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br) |
| 00:06.42 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.156) |
| 00:10.37 | LemensTS | http://pastebin.com/m5a0d4538 having a problem with originate command to DeadAGI app, using phpAGI.........any help? |
| 00:11.23 | LemensTS | i want it to originate to the application deadagi, instead of sending it to the diaplan first |
| 00:12.26 | rue_work | dunno cant help |
| 00:14.50 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 00:16.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 00:19.27 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:23.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:24.03 | dominic1 | there was a manager event for reload in the past |
| 00:24.12 | dominic1 | how can I activate it in 1.6? |
| 00:26.29 | [TK]D-Fender | dominic1: COMMAND <- Same as always |
| 00:27.04 | dominic1 | ? |
| 00:27.10 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 00:27.18 | dominic1 | is there no setting that the eventu is dropped in the manager interface? |
| 00:27.35 | rue_work | I dont know |
| 00:28.14 | [TK]D-Fender | dominic1: ...huh? |
| 00:29.11 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 00:30.16 | path_ | ~books |
| 00:30.55 | [TK]D-Fender | ~book |
| 00:30.56 | 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 |
| 00:31.03 | path_ | thanks :) |
| 00:31.50 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 00:34.25 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 00:36.09 | jaytee | SPOILER ALERT!!!: it's a great book but at the very end they shoot Ole Yeller. |
| 00:36.29 | rue_work | coulda told ya that |
| 00:36.32 | path_ | hahah |
| 00:36.41 | path_ | I'm enjoying it |
| 00:36.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:38.51 | LemensTS | http://www.packtpub.com/asterisk-gateway-interface-programming |
| 00:38.54 | LemensTS | anyone read that |
| 00:39.01 | LemensTS | im gonna order it |
| 00:40.13 | [TK]D-Fender | LemensTS: as a manual for PHPAGI... ok/fine/sure |
| 00:42.06 | rue_work | :) whats this microbrowser thing all about, I managed to get the applications button from "DONT PUSH ME" to just ignoring me |
| 00:42.17 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.132) |
| 00:42.26 | LemensTS | TK: gotta be better than anything else out there. This whole phpagi experience has been rather hard finding information. |
| 00:42.59 | manxpower | AHA! I have found you, you evil \r! |
| 00:43.01 | rue_work | I think I need auser manual |
| 00:43.09 | [TK]D-Fender | LemensTS: the class docs are decent and plenty of code samples out there. |
| 00:44.17 | LemensTS | TK: Yea if they make sense to you. I got more out of reading the actual functions in phpagi.php and phpagi-asmanager.php than anything. |
| 00:47.25 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-209-224.lns10.mel6.internode.on.net) |
| 00:47.46 | keebler | bmoraca: The EZGO's arrived. Taking it apart now. :) |
| 00:49.47 | keebler | Heh |
| 00:49.53 | keebler | Damn thats simple. |
| 00:51.57 | *** join/#asterisk HermesNeto (n=HermesNe@189.71.45.158) |
| 00:52.55 | keebler | Taking pics |
| 00:53.46 | coppice | you bought a golf cart? :-\ |
| 00:54.10 | keebler | http://www.wlanparts.com/product/EZGO-0214/The_EZGo_high_power_outdoor_wireless_client_bridge_24GHz.html |
| 00:54.14 | keebler | Just to test |
| 00:56.03 | iaxy | Hey guys |
| 00:56.55 | iaxy | I'm having trouble with choppy audio and got a call dropped today. Consistant choppy audio. |
| 00:57.28 | iaxy | I have a polycom SIP connected to * and IAX trunk 2 DID's coming in on |
| 00:57.51 | iaxy | I just dropped the payload for the codecs on the polycom down to 10 to see if that will help. |
| 00:57.54 | iaxy | any ideas? |
| 00:58.28 | [TK]D-Fender | iaxy: IAX itself is often an isue |
| 00:58.45 | iaxy | inbound and/or outbound on the IAX trunks doesn't matter.. |
| 00:58.56 | [TK]D-Fender | iaxy: and you have failed to describe what is connected where. |
| 00:58.56 | iaxy | How dare!!! |
| 00:58.59 | drmessano | WHo is the provider? |
| 00:59.25 | iaxy | Polycom SIP connected to * as an extention. |
| 00:59.33 | [TK]D-Fender | iaxy: And lowering your payload size INCREASES your packet rate and overhead waste <- |
| 00:59.35 | drmessano | WHo is the *provider*? |
| 00:59.40 | iaxy | DID from LES.net nailed up with IAX |
| 00:59.46 | [TK]D-Fender | FUCK IAX |
| 00:59.48 | [TK]D-Fender | :p |
| 00:59.50 | drmessano | LES.net IAX = Sux |
| 01:00.01 | drmessano | They use 1.2 and IAX connections are HORRIBLE to them |
| 01:00.02 | drmessano | use SIP |
| 01:00.22 | iaxy | TK, I won't misinterpret that one... |
| 01:01.44 | [TK]D-Fender | iaxy: I see my delicate phrasing has come through loud and clear once more :) |
| 01:02.03 | DJ_HaMsTa | any of u guys having a prob with les.net disconnecting once in a while ? |
| 01:02.03 | harry_v | I thought IAX was the only means to pass though firewalls ? so why give it a bad rap TK? |
| 01:02.07 | iaxy | oh no TK. |
| 01:02.07 | keebler | To those looking for a cheap Wireless Bridge... This one is only $83. Here's what it looks like taken apart. http://img.photobucket.com/albums/v221/Nicca64/IMG00515.jpg?t=1233622877 |
| 01:02.10 | drmessano | IAX works well with never Asterisk and if supported by the provider |
| 01:02.16 | [TK]D-Fender | iaxy: and varying your packet rate between them only causes additional timing issues |
| 01:02.20 | drmessano | SIP works through firewalls just fine |
| 01:02.27 | iaxy | I received those packets and dropped the ones I didn't need... "=-) |
| 01:02.29 | keebler | Based off a Realtek RTL8186 |
| 01:02.41 | drmessano | SIP works just fine thru firewalls when you know what youre doing |
| 01:02.54 | drmessano | IAX implementations by third parties and those using 1.2 suck |
| 01:02.57 | harry_v | drmessano only if you change the firwall to allow rtp/sip to pass though. |
| 01:03.01 | DJ_HaMsTa | drmessano: u have ur did with asterisk from les.net ? |
| 01:03.07 | [TK]D-Fender | harry_v: Bad rap is the host of cases where IAX alone appears to be the culprit for noticable audio issues |
| 01:03.13 | drmessano | harry_v: How the hell does IAX get thru then????? |
| 01:03.17 | [TK]D-Fender | harry_v: Where SIP witht he same provider = perfect |
| 01:03.26 | drmessano | IAX isnt MAGIC |
| 01:03.34 | [TK]D-Fender | harry_v: its a "means well" protocol that is 99% unnecessary |
| 01:03.39 | drmessano | DJ_HaMsTa: Yes |
| 01:03.50 | harry_v | I wanted to demo my system at a church but there firewall was blocking the sip traffic. .. |
| 01:04.06 | DJ_HaMsTa | u just had to configure the sip.conf with their info ? |
| 01:04.12 | drmessano | DJ_HaMsTa: Yeah |
| 01:04.22 | [TK]D-Fender | harry_v: Their firewall? What kind of POS was it? |
| 01:04.34 | harry_v | cisco pix |
| 01:04.46 | drmessano | SIP gets a bad rap.. 99.99% of the time its the USER or shitty firewall |
| 01:04.47 | DJ_HaMsTa | could u send me your sip.conf (remove anything that might be too detailed or personal or confidential) so i can see what it looks like ? |
| 01:04.51 | drmessano | HAHHAHHAHA |
| 01:04.56 | drmessano | Cisco doesnt speak SIP |
| 01:05.07 | harry_v | really |
| 01:05.11 | harry_v | dam |
| 01:05.11 | drmessano | PIX 501 is a HORRIBLE box to put in front of an Asterisk box |
| 01:05.12 | [TK]D-Fender | harry_v: PIX = BLEH, always has and its a known offender |
| 01:05.21 | harry_v | wait, that is church standard. |
| 01:05.22 | [TK]D-Fender | yup |
| 01:05.33 | drmessano | Church Standard?????? |
| 01:05.34 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b4f313a3f51aa96a) |
| 01:05.44 | harry_v | yes, church wide. |
| 01:05.44 | [TK]D-Fender | harry_v: "thats nice" its also "death on wheels" to SIP and sane * setups |
| 01:05.48 | iaxy | You saying they are using * 1.2? |
| 01:05.51 | drmessano | harry_v: Um no |
| 01:05.59 | harry_v | um yes |
| 01:06.24 | drmessano | Really? What about all the churched we service with Sonicwalls and WRT54Gs? |
| 01:06.27 | drmessano | NEW STANDARD? |
| 01:06.31 | harry_v | So, which firewalls are sip friendly? |
| 01:06.31 | drmessano | Whatever man |
| 01:06.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-140a7e6014c5fd2d) |
| 01:06.38 | drmessano | churches |
| 01:06.51 | harry_v | drmessano I was talking about one faith not all of them. |
| 01:06.54 | harry_v | silly |
| 01:06.57 | drmessano | Which faith? |
| 01:07.17 | [TK]D-Fender | drmessano: "I believe SIP should work with this" |
| 01:07.17 | carrar | firewall yourself from EVIL |
| 01:07.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ebcc1f712ccecd0d) |
| 01:07.44 | iaxy | I have had better quality sticking an asterisk box across the country connecting to it via IAX using it as trunk than I am getting from les.net |
| 01:07.48 | harry_v | anyway, need to focus on getting to know which firewalls are sip friendly. IPcop is. not sure about others. |
| 01:07.50 | drmessano | "Roman Catholics love Cisco, but those Zionist.. they are all about some Firebox's" |
| 01:08.02 | drmessano | What crap lol |
| 01:08.15 | carrar | harry_v, I've had PIX's working just fine in front of Asterisk |
| 01:08.24 | harry_v | really |
| 01:08.26 | carrar | yes |
| 01:08.30 | drmessano | So what do the jews and the muslims use? |
| 01:08.37 | carrar | same with NetScreens |
| 01:08.40 | iaxy | Can we focus here..... you have a customer!!!!! |
| 01:08.45 | iaxy | :-) |
| 01:08.52 | drmessano | A customer? |
| 01:08.58 | [TK]D-Fender | iaxy: Stop blaming the provider when we've been beating you over the head that its IAX <- |
| 01:09.09 | [TK]D-Fender | iaxy: So switch protocols and be done with it |
| 01:09.45 | iaxy | I like how you beat around the bush, and drawing out your conclusions TK....:-) |
| 01:09.51 | drmessano | The problem is IAX and 1.2, which is what LES.net happens to be using. It's not Les's fault, it's 1.2 not being so great at IAX |
| 01:09.54 | drmessano | Move on |
| 01:10.03 | drmessano | Use SIP |
| 01:10.08 | drmessano | You wont fix this problem |
| 01:10.12 | [TK]D-Fender | iaxy: I prefer to think of it as "long standing precedence and experience" |
| 01:10.14 | drmessano | IAX is not gonna work here, perioud |
| 01:10.15 | drmessano | IAX is not gonna work here, period |
| 01:10.39 | [TK]D-Fender | iaxy: Esp as I have numerous clients using Les.net who are VERY happy with them, and only ONE guy here with an issue |
| 01:10.42 | iaxy | I believe you guys.... just wanna understand it. you talking * 1.2? |
| 01:10.45 | iaxy | or IAX 1.2 |
| 01:10.49 | harry_v | or put a nat/fw behind a fw that blocks it? |
| 01:11.00 | harry_v | nat/fw sip friendly |
| 01:11.04 | [TK]D-Fender | iaxy: Whose very nick infers a sense of "fanboy-sim" and "denial" |
| 01:11.07 | drmessano | Asterisk 1.2 |
| 01:11.49 | [TK]D-Fender | iaxy: and the IAXY is a dead-end unfriendly little nugget I hope never to have to use. |
| 01:11.50 | drmessano | Asterisk 1.2's IAX implementation was "best effort" at the time, but still flaky. 1.4 and 1.6 are 10x better |
| 01:11.58 | drmessano | Its just not reliable |
| 01:12.06 | sim-Me | I have the original blue one. |
| 01:12.14 | [TK]D-Fender | sim-Me: HORRIBLE |
| 01:12.22 | drmessano | Forget trying to be an elitist little IAX user and use SIP... No one will think you're cool |
| 01:13.00 | drmessano | We all used IAX long before you ever heard of Asterisk, and were 1337 first. |
| 01:13.03 | drmessano | So its been done |
| 01:13.08 | drmessano | Use SIP, make calls, move on |
| 01:13.28 | [TK]D-Fender | drmessano: No, Bob here has in fact been using * for many years... |
| 01:13.40 | [TK]D-Fender | drmessano: Not that the time has worn well upon him :) |
| 01:13.51 | drmessano | Thats sad |
| 01:14.07 | [TK]D-Fender | drmessano: Its a question of focus |
| 01:14.26 | [TK]D-Fender | drmessano: I know nearly jack-sshit about Apache personlly... yet I run it... |
| 01:14.45 | drmessano | [TK]D-Fender: But.. BUT.. BUT.. I want to be cool and use IAX.. I want to be "that guy" |
| 01:14.51 | drmessano | [TK]D-Fender: HELP ME PLZ |
| 01:14.52 | [TK]D-Fender | drmessano: Of course... I make no claims as how best to do so :) |
| 01:15.28 | drmessano | [TK]D-Fender: I MUST.. MUST.. MUUUUST use IAX with my 2 concurrent calls.. I NEED TEH BRANDWITHZ |
| 01:15.45 | [TK]D-Fender | drmessano: That IS the only valid reason for it :) |
| 01:15.48 | drmessano | [TK]D-Fender: I CAN HAZ LOW OVERHED? |
| 01:16.40 | DJ_HaMsTa | whats the diff between DID and SIP ? |
| 01:17.17 | [TK]D-Fender | DJ_HaMsTa: The same as between an airplane and a hamster |
| 01:17.26 | carrar | bwahah |
| 01:17.31 | [TK]D-Fender | ~did |
| 01:17.32 | jbot | hmm... did is Direct Inward Dialing, or just a phone number |
| 01:17.33 | [TK]D-Fender | ~sip |
| 01:17.34 | jbot | somebody said sip was http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 01:17.38 | drmessano | [TK]D-Fender: still amazes me the "I heard IAX could hop a firewall and hotwire a PRI, then hijack a jet to free calls in Kuala Lumpur" argument comes up so often.. Still. |
| 01:17.48 | carrar | I was resisting that one |
| 01:17.54 | [TK]D-Fender | DJ_HaMsTa: 1 is an f-n PHONE NUMBER. and the other is a VoIP call setup protocol :) |
| 01:18.26 | drmessano | IAX is not the answer to NAT and firewall issues. "Not being stupid" is. |
| 01:18.40 | *** join/#asterisk johnakabean (n=none@pool-72-82-108-206.nrflva.east.verizon.net) |
| 01:18.50 | [TK]D-Fender | DJ_HaMsTa: actually it'd be better to compare between a car & the highway. |
| 01:19.25 | [TK]D-Fender | drmessano: I CAN HAZ CISCO PIX PLZ?! |
| 01:19.30 | bmoraca | well...hampsters have been known to power the engines of a jet aircraft... |
| 01:19.33 | carrar | luxury car? |
| 01:19.44 | carrar | mulitlane highway |
| 01:20.52 | johnakabean | asterisk e-mail to callback? |
| 01:21.18 | iaxy | I compiled asterisk befor version 1 |
| 01:21.36 | iaxy | when there was no gui with linux |
| 01:21.41 | bmoraca | and it took you this long to get it working? wow! |
| 01:21.42 | DJ_HaMsTa | highway and car, very good comparison |
| 01:21.55 | iaxy | so i believe your statement is incorrect there. |
| 01:21.57 | drmessano | bmoraca: WIN! |
| 01:21.59 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:22.02 | drmessano | bmoraca: You can stay |
| 01:22.04 | iaxy | I'm slow |
| 01:22.34 | iaxy | I got out for 17 years or so. |
| 01:22.42 | [TK]D-Fender | johnakabean: <TREBEK> I'm sorry you forgot to phrase that in the form of a COMPLETE question. |
| 01:22.49 | iaxy | I need an * box now, so I came back |
| 01:23.18 | johnakabean | i wasjust hinting around to everyone's answers to some question |
| 01:23.21 | bmoraca | i have a moment or two every now and again... |
| 01:23.23 | bmoraca | anyway |
| 01:23.28 | bmoraca | go home time |
| 01:23.30 | jaytee | http://www.osburn.com/asterisk-sign.png |
| 01:23.48 | [TK]D-Fender | jaytee: NEVER "old" :) |
| 01:23.50 | carrar | heh |
| 01:23.59 | jaytee | [TK]D-Fender, a true classic! |
| 01:24.10 | [TK]D-Fender | jaytee: So Day 1 down! Only 1 minor SNAFU and it got fixed.... Samba sticks! |
| 01:24.13 | bmoraca | yay for random pictures |
| 01:24.30 | jaytee | [TK]D-Fender, YAY!!! \o/ |
| 01:24.46 | [TK]D-Fender | jaytee: note : OSX 10.5 + Samba neds a Unix Compat fix or symlinks = BREAKAGE |
| 01:24.56 | johnakabean | I know there are many ways to accomplish a php/perl/etc script to check e-mail every so often and execute a command based on arguments retrieved from the e-mail but what is the best way to have a script check the e-mail and execute the Originate command in asterisk manager. |
| 01:24.59 | [TK]D-Fender | jaytee: Todays quick lesson |
| 01:25.12 | jaytee | [TK]D-Fender, went live on the new IVR today. so far so good. still some speech rec issues when using a cell phone though. screw em! they can still use DTMF |
| 01:25.14 | [TK]D-Fender | johnakabean: That script has nothing to do with * |
| 01:25.16 | carrar | my OSX desktop just froze solid 20 mins ago |
| 01:25.22 | carrar | had to reboot |
| 01:25.58 | [TK]D-Fender | carrar: I watched a MacOSX 10.5 crash screen... crash in mid-draw the other day. CRASH DIFFERENT (tm) |
| 01:26.03 | harry_v | ahh |
| 01:26.18 | carrar | heh |
| 01:26.23 | [TK]D-Fender | carrar: it dragged on at like 5px /s for 1/2 the length and BZZZZZZZZZZ |
| 01:26.24 | johnakabean | originate command is locked inside the asterisk manager; will an argument such as "asterisk originate local/2125551212 extension 1@disa" work? |
| 01:26.33 | carrar | thats messed |
| 01:26.35 | johnakabean | on command line |
| 01:26.36 | *** join/#asterisk xlogik (n=xlogik@c-98-229-61-41.hsd1.ma.comcast.net) |
| 01:26.59 | DJ_HaMsTa | how do i get that web interface in asterisk within ubuntu ? |
| 01:27.18 | jaytee | DJ_HaMsTa, you go to #asterisk-gui and ask there |
| 01:27.34 | [TK]D-Fender | johnakabean: you can call straight from *NIX CLI or a call file, or AMI, yadda yadda |
| 01:27.44 | johnakabean | no, so i'm guessing i have to make the script create a call file |
| 01:28.12 | [TK]D-Fender | johnakabean: Yes, something to the tune of 5-6 lines of plain text. |
| 01:28.22 | [TK]D-Fender | johnakabean: Ain't Raw-Cat Science |
| 01:28.23 | jaytee | there is more than one way to sodomize a cat! er, um, wait! that came out wrong....skin....yeah, that's it, skin a cat. |
| 01:28.47 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 01:28.47 | [TK]D-Fender | jaytee: Is that a bot script, or just phenomenal timing? :) |
| 01:29.04 | johnakabean | i have another php script that makes call files but I am going to have a little trouble having it check e-mail, parsing it, and extracting variables |
| 01:29.05 | jaytee | I'm phenomenal and a legend in my own mind |
| 01:29.22 | [TK]D-Fender | listens closely... |
| 01:29.25 | johnakabean | joins #php |
| 01:29.26 | [TK]D-Fender | jaytee: Hear that? |
| 01:29.39 | [TK]D-Fender | jaytee: Its the sound of noone disagreeing with you ;) |
| 01:29.49 | [TK]D-Fender | *poke*jab* |
| 01:29.52 | jaytee | [TK]D-Fender, hehehe |
| 01:29.59 | *** join/#asterisk ingenius (n=alektro@host253.190-30-205.telecom.net.ar) |
| 01:30.10 | adr|an | [TK]D-Fender : can i ask you a question on private ? |
| 01:30.42 | carrar | Oh My |
| 01:31.00 | [TK]D-Fender | adr|an: I'm skeered |
| 01:31.06 | adr|an | :P))) |
| 01:31.12 | adr|an | don be afraid :) |
| 01:31.22 | jaytee | hopefully it's not about lotions and rubbing |
| 01:31.55 | [TK]D-Fender | hangs a sign over his ass labeled "EXIT ONLY" |
| 01:33.19 | harry_v | Okay, I have a pastebin of my polycom config files. For some reason my ip500 cannot log into the ftpserver. Both serverand phones user/pass match. Any polycom nuts that know there stuff may know what is going on. http://www.pastebin.ca/1325877 |
| 01:33.41 | harry_v | would be helpfull :) |
| 01:33.51 | jaytee | covers his "Live to ride, Ride to Live" tramp stamp tattoo and whistles nonchalantly. |
| 01:34.15 | carrar | can you log in manually? |
| 01:34.20 | harry_v | yes |
| 01:34.27 | carrar | look at the ftp log file? |
| 01:34.32 | carrar | whats it erroring on? |
| 01:34.45 | carrar | may need to enable logging |
| 01:34.50 | harry_v | just on the phones display |
| 01:34.58 | harry_v | cannot find boot server |
| 01:35.13 | carrar | did you try factory reset? |
| 01:35.34 | carrar | by default it should wanting to login as PlcmSpIp |
| 01:35.36 | harry_v | where would vsftpd log files be located. |
| 01:35.38 | drmessano | Um what |
| 01:35.39 | carrar | same pass |
| 01:35.42 | drmessano | [20:22] <iaxy> I'm slow [20:23] <iaxy> I got out for 17 years or so. |
| 01:35.52 | carrar | in /var/log/xferlog |
| 01:35.52 | drmessano | 17 years of Asterisk? |
| 01:35.55 | harry_v | yes, reset to factory and rentered ftp ip/user/pass |
| 01:35.56 | drmessano | wow |
| 01:36.00 | drmessano | EARLY beta tester |
| 01:36.22 | carrar | also |
| 01:36.23 | carrar | # Activate logging of uploads/downloads. |
| 01:36.23 | carrar | xferlog_enable=YES |
| 01:36.27 | carrar | in your vsftp.conf |
| 01:36.46 | [TK]D-Fender | drmessano: Out of Linux... |
| 01:36.57 | [TK]D-Fender | drmessano: I think he was referring to pre 1.0 LINUX, not ASTERISK |
| 01:36.59 | harry_v | okay |
| 01:37.15 | carrar | harry_v, I have also added ftp logging to syslog.conf |
| 01:37.24 | [TK]D-Fender | drmessano: Which equates to "My knowledge is dated... in a CARBON sort of way" |
| 01:37.29 | doug | 17 years of linux, even. how depressing. |
| 01:37.46 | carrar | err no i didn't |
| 01:37.54 | drmessano | [TK]D-Fender: 17 years? So then hes never actually used Linux then.. I mean, 17 years ago it was a kernel and some poo.. Thats like being out of Windows for 25 years. How can you even claim to have used it at this point? |
| 01:38.36 | doug | the majority of the user experience of linux is not really due to linux |
| 01:38.39 | [TK]D-Fender | drmessano: thats what loose associations are for :) |
| 01:38.42 | harry_v | xferlog is empty and xferlog_enable=yes existing in vsftpd.conf file. |
| 01:38.57 | jaytee | Didn't Windows 1.0 ship in 84? Win 3.0 came out in May of 90. |
| 01:39.05 | jaytee | 1.0 was total poo also though |
| 01:39.26 | drmessano | doug: 17 years ago I stared at a cursor. So i'm a little rusty. <--- Overstatement of the year |
| 01:39.26 | doug | i never really used linux until until under a year ago. i've always eschewed it in favor of bsd. |
| 01:39.47 | carrar | harry_v, need to get your ftp server working properly |
| 01:39.56 | [TK]D-Fender | jaytee: I was the shiznit running 2.0 on my XT in EGA mode w/ like... CALCULATOR & NOTEPAD running... whee! |
| 01:39.56 | harry_v | carrer I know. |
| 01:39.57 | harry_v | ;) |
| 01:39.59 | jaytee | I've always had a bit of an overbite so I've avoided eschewing things |
| 01:40.07 | drmessano | I havent really touched Unix since the early part of 71. Have I missed much? |
| 01:40.14 | doug | nothing important. |
| 01:40.48 | [TK]D-Fender | Apparently EXT4 is looking pretty darn good so far <- |
| 01:40.56 | drmessano | My god |
| 01:41.01 | drmessano | What happened to EXT2 ? |
| 01:41.05 | drmessano | or 3? |
| 01:41.18 | jaytee | [TK]D-Fender, ooooooh, 16 colors!!! I would have been so jealous and probably would have tried to sell my Hercules adapter and some porn to buy an EGA card and monitor |
| 01:41.26 | drmessano | Wait, wait |
| 01:41.29 | drmessano | COLORS? |
| 01:41.46 | harry_v | carrer, take a look at this and tell me how accurate it is. Wife needs meto go some where. |
| 01:41.51 | [TK]D-Fender | drmessano: 2 = Dodo, 3 = mainstream.... but we all know that ReiserFS is a KILLER file-system :) |
| 01:41.58 | harry_v | if you dont mind. Then will follow its instructions. |
| 01:42.02 | harry_v | http://www.sureteq.com/asterisk/polycom.htm#4.%C2%A0_FTP_configuration_ |
| 01:42.09 | [TK]D-Fender | jaytee: Herc had an equivalent to EGA in the day... |
| 01:42.15 | [TK]D-Fender | jaytee: And higher performing too |
| 01:42.32 | drmessano | ReiserFS butchered the competition |
| 01:42.42 | harry_v | and in prison |
| 01:42.44 | drmessano | Was better at hiding files |
| 01:42.53 | [TK]D-Fender | drmessano: Lots of fragmentation... |
| 01:43.25 | drmessano | harry_v: Are you the obvious guy that ruins the subtlety of the joke by telling us "You know hes in prison, right?" |
| 01:43.25 | jaytee | I remember calling Packard Bell to ask how to program the "Macro" key on my 105 key keyboard. "Um, that doesn't do anything. We just got a large shipment of them but they couldn't get it to work with the system bios or something." |
| 01:43.35 | carrar | harry_v, doesn't say how they are running vsftp |
| 01:43.49 | carrar | out of inetd or stand alone |
| 01:44.01 | harry_v | I see |
| 01:44.57 | carrar | probably stand alone |
| 01:45.07 | carrar | But get logging working |
| 01:45.14 | carrar | So you can see what your phone is doing |
| 01:45.18 | jaytee | Alas! gone are the fun days of having someone ask you, "What's the difference between expanded memory and extended memory and after giving them a detailed 15 minute explanation they've fallen asleep and then wake up later with their wallet missing. |
| 01:48.12 | [TK]D-Fender | jaytee: pwned |
| 01:48.40 | [TK]D-Fender | jaytee: I remember my full-length, full height2mb ISA expansion boards |
| 01:49.19 | [TK]D-Fender | jaytee: Expanded was such a rip.... mostly go used for a friggen ram-disk :) |
| 01:49.46 | jaytee | [TK]D-Fender, my 286/12mhz system had an expansion slot but the manufacture discontinued the proprietary card because of poor sales. |
| 01:50.01 | jaytee | expanded was truly a waste |
| 01:53.12 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 01:53.55 | jaytee | bbiab |
| 01:54.08 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-7605852c4da30c68) |
| 01:56.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 01:59.27 | *** join/#asterisk stabler (n=seedbox@rrcs-70-60-8-130.central.biz.rr.com) |
| 01:59.46 | stabler | can anyone help with installing sccp on my asterisk server |
| 02:00.40 | stabler | how do i determine wheather i have asterisk version v1_0 or HEAD |
| 02:02.30 | *** join/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com) |
| 02:02.48 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:03.03 | Qwell | stabler: Any version information you have is extremely out of date. |
| 02:03.34 | Gopher_77 | I'm trying to get * working on my linux machine. My dahdi devices are in use by *, but from there I don't know how to configure it. Can I get some help? |
| 02:04.11 | stabler | so what do i input for "ASTERISK_VERSION=" |
| 02:04.37 | Qwell | stabler: Neither. Those instructions are severely out of date. |
| 02:04.59 | stabler | so i need to find a newer version of sccp |
| 02:05.06 | Qwell | stabler: You need to either find more recent instructions, or use chan_skinny |
| 02:05.44 | stabler | will chan_skinny support my cisco phone just as well as sccp? |
| 02:05.56 | Qwell | Far better |
| 02:06.00 | stabler | i have a cisco 7940 ip phone |
| 02:07.17 | stabler | Qwell, thanks for the info |
| 02:07.55 | stabler | i cant convert the mofo to sip so i have to do it like this |
| 02:08.19 | [TK]D-Fender | Gopher_77: It configures much like zaptel. Go read the docs on setting up Zaptel then go read the docs on DAHDI in the tarball |
| 02:10.59 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 02:14.30 | stabler | is chan_sccp a better option? |
| 02:14.40 | stabler | oops |
| 02:14.41 | stabler | nevm |
| 02:14.43 | stabler | *nvm |
| 02:15.16 | Gopher_77 | [TK]D-Fender: I've configured the driver, and the devices are in use by * when I do lsdahdi |
| 02:16.03 | [TK]D-Fender | Gopher_77: Pastebin something useful and show us the problem |
| 02:16.31 | Gopher_77 | [TK]D-Fender: what's useful? And the problem is that I don't have a dial tone and dialing seems to do nothing |
| 02:17.19 | [TK]D-Fender | Gopher_77: Problem is you aren't showing us your configs or that your kernel module is properly loaded, or dmesg output to check for warning or ANYTHING |
| 02:17.53 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 02:18.09 | Gopher_77 | [TK]D-Fender: which configs are useful? There are several. |
| 02:18.50 | [TK]D-Fender | Gopher_77: Do you even have to ask? How about everything that TOUCHES the DHADI subsystem to start... |
| 02:21.44 | stabler | looks like chan_sccp_b is a good option for me |
| 02:22.00 | stabler | does anyone have any insite on chan_sccp_b |
| 02:22.48 | Gopher_77 | [TK]D-Fender: http://nopaste.com/p/aCmKZn9Pw |
| 02:23.24 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 02:24.54 | [TK]D-Fender | waits for the rest... |
| 02:25.07 | stabler | lol |
| 02:25.10 | Gopher_77 | [TK]D-Fender: what's the rest? |
| 02:25.26 | [TK]D-Fender | Gopher_77: You only provided HALF of the config files for DAHDI |
| 02:25.49 | Gopher_77 | [TK]D-Fender: I don't know about anything else; maybe that's the problem |
| 02:26.01 | [TK]D-Fender | Gopher_77: Apparently |
| 02:26.22 | [TK]D-Fender | Gopher_77: Because you have not defined 1 channel for * to use there yet. |
| 02:26.29 | Gopher_77 | [TK]D-Fender: init.conf? it's all comments |
| 02:26.34 | [TK]D-Fender | Gopher_77: chan-dahdi.conf <------------------ |
| 02:26.43 | Gopher_77 | [TK]D-Fender: oh yeah |
| 02:29.02 | *** join/#asterisk bmoraca (n=bmoraca@adsl-75-12-126-173.dsl.skt2ca.sbcglobal.net) |
| 02:29.11 | [TK]D-Fender | Gopher_77: And no confirmation for "dahdi_cfg -vvvv" to prove the module loaded ok. No interrupt dump to show the .ko is loaded.... |
| 02:29.16 | [TK]D-Fender | waits some more... |
| 02:31.06 | Gopher_77 | [TK]D-Fender: http://nopaste.com/p/arrNLqI2b |
| 02:32.25 | [TK]D-Fender | Gopher_77: *chan-dahdi.conf* <-- not a single channel defined in there |
| 02:36.31 | *** join/#asterisk Khratos (n=Khratos@190.166.130.247) |
| 02:39.47 | Gopher_77 | [TK]D-Fender: oops, I grepped wrong, repasted: http://nopaste.com/p/akAOgML1H |
| 02:41.15 | [TK]D-Fender | Gopher_77: Channel 4 not configured. next go look from * CLI and see if chan_dahdi.so is loaded. then reload it |
| 02:42.25 | phix | hmmmm, TDM400p's, they support pass through on no power? |
| 02:43.09 | bmoraca | phix: i do not believe so |
| 02:46.07 | [TK]D-Fender | phix: No such thing as "passthrough |
| 02:46.19 | [TK]D-Fender | phix: Powered, or otherwise |
| 02:46.38 | phix | there should be an option to set which modules to pass through :) |
| 02:46.40 | Gopher_77 | http://nopaste.com/p/alOGpquLB |
| 02:46.50 | phix | via jumpers or DIP switches or something |
| 02:46.51 | phix | oh well |
| 02:46.55 | [TK]D-Fender | phix: Feel free to start soldering. |
| 02:47.43 | [TK]D-Fender | Gopher_77: Show me your attempt to unload chan_dahdi and reload it |
| 02:48.05 | phix | [TK]D-Fender: :) |
| 02:48.22 | phix | [TK]D-Fender: sure, I will just get the circuit diagram from digium first.... |
| 02:49.23 | Gopher_77 | [TK]D-Fender: I don't see in the help how to do that... can you give me the command? |
| 02:49.38 | Gopher_77 | [TK]D-Fender: this is my first time using *CLI |
| 02:49.43 | [TK]D-Fender | phix: Other makers have made power-failover modules. |
| 02:50.02 | [TK]D-Fender | Gopher_77: "module unload chan_dahdi.so" |
| 02:50.07 | [TK]D-Fender | Gopher_77: "module load chan_dahdi.so" |
| 02:50.47 | Gopher_77 | failed |
| 02:51.08 | phix | [TK]D-Fender: sweet |
| 02:51.12 | Gopher_77 | oops never mind |
| 02:51.36 | Gopher_77 | [TK]D-Fender: http://nopaste.com/p/akDorFkr9 |
| 02:52.14 | [TK]D-Fender | Gopher_77: Set verbose 10 |
| 02:52.28 | [TK]D-Fender | Gopher_77: And check your channels in between |
| 02:53.58 | Gopher_77 | [TK]D-Fender: how do I set verbose 10? |
| 02:54.13 | [TK]D-Fender | Gopher_77: "core set verbose 10" |
| 02:54.19 | phix | [TK]D-Fender: are they cheap |
| 02:54.35 | Gopher_77 | ah, ok |
| 02:54.53 | *** join/#asterisk tjz (n=tjz@bb121-7-26-157.singnet.com.sg) |
| 02:55.40 | Gopher_77 | [TK]D-Fender: http://nopaste.com/p/a3cc9T0G6 |
| 02:57.01 | *** join/#asterisk keebler (n=keebler@h20.148.20.98.dynamic.ip.windstream.net) |
| 02:58.30 | [TK]D-Fender | Gopher_77: I said between loading and unloading the module... |
| 02:58.56 | [TK]D-Fender | phix: where cheap = woudn't touch witha 10' pole |
| 02:59.54 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 03:01.08 | iaxy | WOOHOO! |
| 03:01.46 | Gopher_77 | [TK]D-Fender: http://nopaste.com/p/aTEbPyLgC |
| 03:02.23 | [TK]D-Fender | Gopher_77: ok, go try and take a phone off-hook |
| 03:02.57 | Gopher_77 | [TK]D-Fender: no dial tone, but I hear when I rub the microphone |
| 03:03.28 | [TK]D-Fender | Gopher_77: check all the ports. |
| 03:03.44 | [TK]D-Fender | Gopher_77: check CLI to see if it registers |
| 03:04.12 | sipy | TK.... thanks man, sip works much better. |
| 03:04.21 | sipy | glad I thought of that...:-) |
| 03:04.32 | [TK]D-Fender | ~cluebat sipy |
| 03:04.33 | jbot | ACTION pulls out a ClueBat (tm) and thwaps sipy. |
| 03:04.44 | [TK]D-Fender | ClueBat (tm) NEVER MISSES!!!!!!!!!!!! |
| 03:04.51 | sipy | haha |
| 03:04.59 | Gopher_77 | [TK]D-Fender: no dial tone in any, but 2 I hear the microphone rub, and 2 I don't |
| 03:05.05 | [TK]D-Fender | sipy: Next time... just f'n listen to us, ok? :p |
| 03:05.20 | sipy | I did, thats why I'm dancing! |
| 03:05.43 | [TK]D-Fender | Gopher_77: Try another phone. Do you see anything in CLI when you pick up? |
| 03:05.51 | Gopher_77 | needs a ClueBat (tm) |
| 03:06.17 | [TK]D-Fender | ~cluebat Gopher_77 |
| 03:06.18 | jbot | ACTION pulls out a ClueBat (tm) and thwaps Gopher_77. |
| 03:06.24 | [TK]D-Fender | ClueBat (tm) NEVER MISSES!!!!!!!!!!!! |
| 03:06.30 | [TK]D-Fender | NEXT!!@@!!@! (c) BKW |
| 03:06.32 | Gopher_77 | [TK]D-Fender: oh yeah, it's been registering my activity |
| 03:06.47 | [TK]D-Fender | Gopher_77: Shoe |
| 03:06.50 | [TK]D-Fender | show* |
| 03:07.10 | Gopher_77 | <PROTECTED> |
| 03:07.10 | Gopher_77 | <PROTECTED> |
| 03:08.20 | [TK]D-Fender | Gopher_77: is the hangup instant or when you actually hangup? |
| 03:08.46 | Gopher_77 | [TK]D-Fender: when I hang up, and it is going active when I pick up (checked show in the middle) |
| 03:09.01 | [TK]D-Fender | Gopher_77: Ok, PB your dialplan |
| 03:09.28 | *** join/#asterisk adr|an (n=xpl@unaffiliated/adrianxxx) |
| 03:09.35 | Gopher_77 | [TK]D-Fender: ok I think this is where I need my clue |
| 03:09.59 | Gopher_77 | [TK]D-Fender: http://nopaste.com/p/alZKz5oCR |
| 03:10.35 | [TK]D-Fender | Gopher_77: Those contexts don't match your chan-dahdi.conf |
| 03:11.16 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 03:11.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:11.17 | [TK]D-Fender | Gopher_77: You HAVE nothing you can dial |
| 03:11.53 | Gopher_77 | [TK]D-Fender: doesn't surprise me |
| 03:13.09 | Gopher_77 | [TK]D-Fender: I have a phone and a fax machine I can hook up |
| 03:13.50 | [TK]D-Fender | Gopher_77: Just use your regular phone you were testing before and fix your dialplan. You have nothing you can dial. Good reason for * to bitch at you |
| 03:17.37 | Gopher_77 | [TK]D-Fender: how do I reference a Dahdi device? I think zaptel was like Zap/1 |
| 03:17.41 | Gopher_77 | [TK]D-Fender: right? |
| 03:18.05 | [TK]D-Fender | Gopher_77: "dahdi/1", etc |
| 03:18.28 | [TK]D-Fender | Gopher_77: Not important yet though. Just sat up a dummy exten with a big pattern rage. |
| 03:18.34 | [TK]D-Fender | range* |
| 03:19.24 | Gopher_77 | [TK]D-Fender: it's all greek to me |
| 03:20.17 | [TK]D-Fender | Gopher_77: exten => _xxxx,1,NoOp(Entered 4-digits "${EXTEN}") |
| 03:24.01 | *** join/#asterisk colinm_ (n=colinmat@VDSL-130-13-98-211.PHNX.QWEST.NET) |
| 03:24.44 | Gopher_77 | [TK]D-Fender: which context do I use? |
| 03:25.02 | [TK]D-Fender | Gopher_77: How about the one you told your CHANNEL to use... |
| 03:25.46 | Gopher_77 | [TK]D-Fender: from-internal? |
| 03:26.08 | [TK]D-Fender | Gopher_77: Can't read your own config files? |
| 03:26.13 | Gopher_77 | [TK]D-Fender: nope |
| 03:26.24 | Gopher_77 | [TK]D-Fender: I don't understand contexts |
| 03:26.30 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 03:26.48 | [TK]D-Fender | Gopher_77: yes... [from-internal] |
| 03:26.54 | Gopher_77 | told [TK]D-Fender he needs a ClueBat (tm) |
| 03:27.09 | [TK]D-Fender | Gopher_77: A whole LOT of it |
| 03:27.47 | Gopher_77 | [TK]D-Fender: so when I pick up, it starts with instructions at the context that I set in chan_dahdi.conf? |
| 03:28.24 | [TK]D-Fender | Gopher_77: If it has nothing it can possibly dial then perhaps it won't even give you a dialtone |
| 03:29.06 | Gopher_77 | [TK]D-Fender: perhaps |
| 03:30.02 | sipy | 2nd trunk transfered to SIP. FAN-mofo_TABULOUS |
| 03:31.40 | Gopher_77 | [TK]D-Fender: dialplan repaste > http://nopaste.com/p/aXebEd5Hw |
| 03:32.13 | Gopher_77 | [TK]D-Fender: getting a ring on 4002 |
| 03:32.23 | [TK]D-Fender | Gopher_77: reload your configs and test. |
| 03:32.42 | Gopher_77 | [TK]D-Fender: sounded like my fax machine |
| 03:32.52 | sipy | WTF is channel gahdi? |
| 03:33.03 | Gopher_77 | [TK]D-Fender: no, fax on 4001 |
| 03:33.19 | [TK]D-Fender | sipy: chan_ghandi is pease & quiet |
| 03:33.23 | [TK]D-Fender | peace* |
| 03:33.51 | [TK]D-Fender | Gopher_77: I only asked if you go DIALTONE or if * reacted to what you dialed/. |
| 03:35.08 | sipy | haha |
| 03:35.39 | sipy | and what are you guys confiburlating? |
| 03:35.43 | stabler | what is the pastebin address? |
| 03:40.20 | Gopher_77 | [TK]D-Fender: oh, I had no dial tone but * was putting my call through |
| 03:40.42 | [TK]D-Fender | Gopher_77: Do you have a proper indications.conf? |
| 03:42.34 | Gopher_77 | [TK]D-Fender: default, and it's big |
| 03:43.07 | [TK]D-Fender | Gopher_77: From here, check with Digium support |
| 03:43.34 | Gopher_77 | [TK]D-Fender: will they support * on an openvox card? |
| 03:43.43 | russellb | no. |
| 03:43.45 | stabler | [TK]D-Fender, what is the address for the pastebin |
| 03:44.09 | [TK]D-Fender | stabler: There have been a #&$^ HUNDRED of them linked in the last hour. |
| 03:44.28 | [TK]D-Fender | stabler: and I STILL see on on scrren NOW |
| 03:44.37 | russellb | jbot: tell stabler about pb |
| 03:44.49 | frogonwheels | Now that was weird. Did an attended transfer (flash on a handset connected to a pap2t) - the other end answered, but the person wasn't available, so I pressed flash again.. the person came back, but MOH was still active! |
| 03:45.06 | [TK]D-Fender | grabs his ClueBat (tm) again... gonna be a LONG night |
| 03:45.16 | frogonwheels | [TK]D-Fender: ah yeah. |
| 03:45.32 | *** part/#asterisk Khratos (n=Khratos@190.166.130.247) |
| 03:45.33 | frogonwheels | [TK]D-Fender: point me at something which explains that .. ppplease |
| 03:45.37 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 03:46.26 | jaytee | "Life is like a box of chocolates, ya nevah know what yur gonna git." |
| 03:46.54 | jaytee | ~pb |
| 03:46.55 | 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/ |
| 03:47.18 | [TK]D-Fender | frogonwheels: 3-way call <- |
| 03:47.26 | carrar | jaytee, how many boxes of chocolates have you ever had where you did not know what you were purchasing inside |
| 03:47.30 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:47.39 | carrar | It says right on the BOX!! |
| 03:47.39 | jaytee | carrar, none |
| 03:48.06 | carrar | "Life is like a box of chocolates, if you open your eyes you will know what to expect" |
| 03:48.25 | [TK]D-Fender | hands carrar a spoon. |
| 03:48.34 | [TK]D-Fender | carrar: If thine eye offends thee.... |
| 03:48.42 | [TK]D-Fender | takes the spoon back |
| 03:48.48 | carrar | I bent it anyways |
| 03:48.50 | [TK]D-Fender | hands carrar a rusty spork |
| 03:49.13 | carrar | There is no spork! |
| 03:49.15 | stabler | ok |
| 03:49.17 | *** join/#asterisk hadi- (n=Hadi@CPE002129717ae3-CM001a668ee8b2.cpe.net.cable.rogers.com) |
| 03:49.19 | hadi- | hello |
| 03:49.20 | carrar | unless we are at TachHell |
| 03:49.25 | carrar | Taco |
| 03:49.30 | stabler | im having some issues with chan_sccp_b |
| 03:49.31 | stabler | http://www.nopaste.com/p/aGbV97W7y |
| 03:49.44 | stabler | there is the output for asterisk console and my sccp.conf file |
| 03:49.59 | hadi- | can someone tell me please what file I need to edit in Asterisk 1.4 to change the Music on Hold for Parked calls from default to something else?! |
| 03:49.59 | stabler | my phone says asterisk connected then reboots |
| 03:50.15 | stabler | phone=cisco 7940 |
| 03:50.21 | [TK]D-Fender | hadi-: its based on the the class of the channel that parked it |
| 03:50.51 | *** join/#asterisk icel (n=dan@75.146.143.126) |
| 03:50.58 | hadi- | [TK]D-Fender: This is the setup: Call comes in from DID -> call queue -> call is parked |
| 03:51.08 | carrar | Set(CHANNEL(musicclass)=default) |
| 03:51.27 | hadi- | im only interested in changing the class for parked calls |
| 03:51.29 | hadi- | is this possible? |
| 03:51.29 | [TK]D-Fender | hadi-: FreePBX is NOT supported here. Fix the class of the channel it comes in on yourself |
| 03:51.42 | frogonwheels | [TK]D-Fender: ok. I can see how it's a 3-way call. I'm just not expecting to hear the other person as well as the MOH - and I'm not sure what triggered it. |
| 03:52.05 | [TK]D-Fender | frogonwheels: I see nothing and have even less to add at this point. |
| 03:52.07 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 03:53.01 | frogonwheels | [TK]D-Fender: ok. so how should I have gone about getting the person back without MOH accompanying them,. |
| 03:53.26 | [TK]D-Fender | frogonwheels: I'm hearing a spotty description and seeing nothing. |
| 03:53.46 | [TK]D-Fender | frogonwheels: And notnecessarily trusting the recounting of the chain of events. |
| 03:55.12 | frogonwheels | [TK]D-Fender: Ok. I got a call from A. I pressed flash.. and I dialed another (external) number (ok.. missed that step).. somebody answered, but not the correct person. So I pressed flash again. |
| 03:55.36 | [TK]D-Fender | still see nothing. |
| 03:55.39 | frogonwheels | [TK]D-Fender: That left me talking to A again, but with MOH still on as well |
| 03:56.02 | frogonwheels | [TK]D-Fender: as in we could both here each other as well as the MOH... |
| 03:56.08 | frogonwheels | [TK]D-Fender: I wasn't expecting the MOH. |
| 03:56.09 | [TK]D-Fender | yawns |
| 03:56.21 | frogonwheels | [TK]D-Fender: so you think this is expected behaviour? |
| 03:56.34 | jaytee | "I wasn't expecting the Spanish Inquisition!" |
| 03:56.43 | [TK]D-Fender | frogonwheels: Yes... I fully expect you to keep on rambling :) |
| 03:57.31 | frogonwheels | [TK]D-Fender: Sometimes you can be a real pain, y'know. |
| 03:57.33 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 03:57.43 | aiksa[LV] | hi everybody. |
| 03:57.55 | frogonwheels | jaytee: nope - the Spanish Inquisition would have been equally unexpected indeed :) |
| 03:58.18 | aiksa[LV] | where does "framein: no samples for alawtolin" comes from? transcoding problems? |
| 03:58.28 | [TK]D-Fender | frogonwheels: I see you're having trouble with what I've been telling you |
| 03:58.32 | aiksa[LV] | jaytee: hi, how did it go with your sip trunk? |
| 03:58.46 | [TK]D-Fender | frogonwheels: Its a common issue really. |
| 03:58.53 | jaytee | aiksa[LV], I rolled back to using AIX for now |
| 03:59.03 | [TK]D-Fender | frogonwheels: Chronic fo some, less for others.... |
| 03:59.08 | aiksa[LV] | fromuser didnt help then? |
| 03:59.11 | [TK]D-Fender | frogonwheels: You should know better. |
| 03:59.14 | frogonwheels | [TK]D-Fender: you've said "3-way call" |
| 03:59.27 | jaytee | aiksa[LV], really didn't have time to mess with it today. |
| 03:59.28 | [TK]D-Fender | [22:53]<[TK]D-Fender>frogonwheels: I'm hearing a spotty description and seeing nothing. |
| 03:59.53 | frogonwheels | [TK]D-Fender: I've explained exactly what happened. I don't _believe_ any config files are involved.. |
| 03:59.54 | aiksa[LV] | jaytee: I took a look at my setup and what I did was simply to allow all of the traffic from a specific IP |
| 03:59.56 | [TK]D-Fender | frogonwheels: WHERE'S MY FUCKING PASTEBIN? :P |
| 04:00.47 | [TK]D-Fender | frogonwheels: I don't trust what I don't see, and the bigger story you spin for me the less I care. |
| 04:00.47 | [TK]D-Fender | frogonwheels: "Show me the money" - Jerry McGuire |
| 04:00.47 | aiksa[LV] | not a nice solution, but the network is internal only, so no big deal |
| 04:00.50 | [TK]D-Fender | *sigh* |
| 04:01.23 | jaytee | frogonwheels, most people like to see what they're trying to help someone with and the more the person who asks for help keeps jerking them around the more they just WANT TO RIP YOUR HEAD OFF AND SHIT DOWN YOUR NECK!!! |
| 04:01.46 | [TK]D-Fender | jaytee: No... I usually stop at dismemberment :) |
| 04:02.06 | [TK]D-Fender | jaytee: Noone is worth dedicating digestive processes on following :) |
| 04:02.12 | jaytee | [TK]D-Fender, I'm talking me! I know you have more restraint |
| 04:02.29 | [TK]D-Fender | jaytee: Yeah... I'll only kill :) |
| 04:02.31 | drmessano | grabs his neck stretcher |
| 04:02.34 | ricko73 | [TK]D-Fender: did you invent a silent 10G POE switch that can charge a Tesla? |
| 04:02.55 | drmessano | Did someone say newb? |
| 04:02.59 | drmessano | perks up |
| 04:03.04 | [TK]D-Fender | hooks up his Mr. Fusion to a 5ess switch and FRIES ricko73 |
| 04:03.10 | ricko73 | lol |
| 04:03.14 | ricko73 | evening |
| 04:03.23 | drmessano | creates a final dialPLAN for ricko73 |
| 04:03.32 | ricko73 | hey now |
| 04:03.32 | stabler | im having some issues with chan_sccp_b http://www.nopaste.com/p/aGbV97W7y |
| 04:03.33 | stabler | there is the output for asterisk console and my sccp.conf file |
| 04:03.34 | jaytee | it's just after watching the back and forth and realizing he's never gonna give up the real info and just keep jerking everyone's chain I find myself wanting to reach through the screen and grab his scrawny little chicken neck and scream, "GIVE UP THE PASTEBIN NOW, BIATCH!!!" |
| 04:04.06 | stabler | my phone says asterisk connected.. then reboots |
| 04:04.14 | stabler | phone = cisco 7940 |
| 04:04.22 | jaytee | I finally made over 1 billion dollars in Mafia Wars on Facebook today and now have lost all interest. |
| 04:04.29 | stabler | am i missing something obvious in my config? |
| 04:05.08 | harry_v | mafia wars? |
| 04:05.24 | jaytee | it's a game on Facebook and other social net sites |
| 04:05.33 | harry_v | interesting |
| 04:05.37 | jaytee | it is |
| 04:05.58 | drmessano | jaytee: Asterisk wars is much cooler |
| 04:06.10 | aiksa[LV] | I`ll chime once again with my question: where this is comming from "[Feb 3 05:48:56] WARNING[32424]: translate.c:175 framein: no samples for alawtolin" |
| 04:06.12 | drmessano | jaytee: I shut down and newb and got a free TDM410 card |
| 04:06.18 | jaytee | one rival mobster I fought earlier actually has a horse's head in his collection. |
| 04:06.19 | drmessano | a/and/a/ |
| 04:06.22 | drmessano | s/and/a/ |
| 04:06.24 | drmessano | GRRRR |
| 04:06.32 | drmessano | jaytee: I shut down a newb and got a free TDM410 card |
| 04:06.37 | jaytee | hahaha |
| 04:06.39 | drmessano | It was teh awesum |
| 04:06.53 | harry_v | I think setting up a virtual world seen though eyeglass HUD and the world changes when you move in relation to a gps reciver on your hemet would be much more interesting. |
| 04:07.04 | drmessano | I got someone to pastebin a config, and I [TK]D-Fenders head spun |
| 04:07.07 | jaytee | "You lose! your consolation prize is a Grandstream GXP-2000 and an X100P card! |
| 04:07.11 | aiksa[LV] | should I rebuild samples? I remember asterisk doing that kind of thing upon make install sequence. |
| 04:07.21 | aiksa[LV] | jaytee: :)) |
| 04:07.32 | jaytee | aiksa[LV], that'll overwrite any config you have |
| 04:07.37 | drmessano | jaytee: I beat the end guy in level 2 by stabbing him with an X100P |
| 04:08.07 | jaytee | the edges of the those cards are sharp, sharper than the engineers minds who designed them. |
| 04:08.15 | aiksa[LV] | jaytee: i guess that a quick look at the Makefile should point me to the right command for building those files |
| 04:08.27 | frogonwheels | [TK]D-Fender: http://pastebin.com/d414eb8f3 (btw.. 'pb of your log' would have been sufficient) |
| 04:08.35 | aiksa[LV] | I am just not sure that I need to rebuild them |
| 04:08.58 | jaytee | why would you need to rebuild sample config files? |
| 04:09.29 | drmessano | jaytee: In level 3, if you compile TRUNK and install dahdi and zaptel, your trixbox explodes and wipes out the newbmaster |
| 04:09.50 | aiksa[LV] | jaytee, not the sample config files. |
| 04:09.54 | aiksa[LV] | audio samples |
| 04:10.15 | aiksa[LV] | I am trying to understand where did this comes from: framein: no samples |
| 04:10.15 | aiksa[LV] | <PROTECTED> |
| 04:10.19 | jaytee | ah, you mean your audio files for prompts and such? |
| 04:10.46 | jaytee | no alaw formatted sound files |
| 04:10.49 | aiksa[LV] | jaytee: I distantly remember that upon installing asterisk is making audio sample files for transcoding |
| 04:10.59 | aiksa[LV] | I could be off by a mile here |
| 04:11.12 | jaytee | it's an option in make menuselect to select additional audio file formats |
| 04:11.22 | aiksa[LV] | jaytee: I dont think this is the problem with prompts here |
| 04:11.42 | aiksa[LV] | I get this message on early audio from a telco |
| 04:11.51 | drmessano | WTF |
| 04:12.18 | *** join/#asterisk DarkRift (n=dark@65.92.170.205) |
| 04:12.49 | aiksa[LV] | I "suppose" it could be due to asterisk not being able to transcode that material, or part of it. |
| 04:13.05 | aiksa[LV] | again I could be mile away from the true cause. |
| 04:13.31 | jaytee | if you don't have the codecs for alaw and slin then it might be a problem :-) |
| 04:13.51 | aiksa[LV] | It doesnt have any effect on the whole system, but still I dont like Warning messages in my CLI output. |
| 04:14.13 | aiksa[LV] | jaytee: if I didnt have the codecs I wouldnt be able to hear that message at all i suppose |
| 04:16.01 | aiksa[LV] | oh well I was wrong i dont have anything of a resemblance to codec_slin loaded |
| 04:16.12 | aiksa[LV] | much simplier than I thought. stupid me ...:P |
| 04:16.18 | [TK]D-Fender | frogonwheels: SIP debug helps... as well as seeing the entire call. |
| 04:17.21 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 04:18.24 | aiksa[LV] | hmm, [TK]D-Fender I just googled upon your conversation half a year ago with somebody where you stated that slin is an inbuilt codec for asterisk |
| 04:19.17 | frogonwheels | [TK]D-Fender: that was it except for a couple of dialog destroys I missed. |
| 04:19.37 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 04:20.25 | stabler | can anyone help me out with chan_sccp? |
| 04:20.35 | [TK]D-Fender | frogonwheels: There was no debug for any of the calls |
| 04:20.46 | aiksa[LV] | stabler: sorry - I dont use it |
| 04:20.48 | [TK]D-Fender | stabler: Few use it I'd try again in a few hours |
| 04:21.49 | *** join/#asterisk mrsci (n=sq@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
| 04:21.58 | frogonwheels | [TK]D-Fender: ok. It'll take a bit to reproduce. I'll do it with a sip set debug |
| 04:22.11 | frogonwheels | [TK]D-Fender: I'll leave it for now. |
| 04:22.29 | [TK]D-Fender | frogonwheels: ok/fine/sure |
| 04:23.38 | hadi- | [TK]D-Fender: so how do we change this exactly? |
| 04:23.50 | [TK]D-Fender | ~freepbx |
| 04:23.51 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 04:23.55 | [TK]D-Fender | ^^^^^^^^^^^ |
| 04:25.40 | LemensTS | hey TK, you said you understood the phpAGI classes, Originate (string $channel, [string $exten = NULL], [string $context = NULL], [string $priority = NULL], [string $application = NULL], [string $data = NULL]) How do i do that Originate cmd for deadagi and test.agi as the $data? I still haven't figured out how to leave $exten, $context, $priority as empty strings... |
| 04:26.45 | *** join/#asterisk heison (n=heison@i209-195-80-5.cia.com) |
| 04:26.46 | aiksa[LV] | _does not parse_ |
| 04:27.05 | hadi- | we are not running freebsd |
| 04:27.09 | hadi- | I just need to know |
| 04:27.21 | hadi- | how to change the category for moh in the parking lot |
| 04:27.25 | LemensTS | TK: huh i just tried it and now it works |
| 04:27.39 | LemensTS | been trying it off and on for 3 days |
| 04:28.09 | LemensTS | wow that completes incredibly faster than going thru the dialplan to initiate DeadAGI |
| 04:28.25 | sipy | add parkedmusicclass=default to whatever cat you want., to features_general_something.conf |
| 04:28.28 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:30.11 | aiksa[LV] | what would be the correct cause code to indicate that I am not able to forward the call because all of the lines are taken? 34? |
| 04:30.30 | [TK]D-Fender | [23:30]===hadi-: member of #asterisk and #freepbx |
| 04:30.44 | [TK]D-Fender | [20:56]<Hadi>anyone here know how to change the moh for all parked calls from Category: default to anther category? |
| 04:31.11 | [TK]D-Fender | hadi-: Who do you think you're kidding? |
| 04:31.49 | hadi- | and why do you assume its a freepbx im asking about |
| 04:31.56 | hadi- | only because im in the channel? |
| 04:32.01 | [TK]D-Fender | asking in their channel, then here.... |
| 04:32.33 | [TK]D-Fender | hadEither way I told you its in the device setup of the channel that parked them. |
| 04:34.28 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 04:36.50 | *** join/#asterisk mbt (n=mbt@zest.spicerack.trausch.us) |
| 04:37.55 | mbt | Quick question: Is Asterisk supported on BSD systems or only on GNU/Linux systems? |
| 04:38.44 | frogonwheels | mbt: http://www.voip-info.org/wiki/view/Asterisk+FreeBSD |
| 04:39.41 | mbt | Yeah, I'd fallen across that, though it seems to indicate that it may not be production stable. Then again, it's from late 2007, according to the header, which is why I was asking to be sure. |
| 04:39.56 | frogonwheels | shrugs. |
| 04:40.18 | mbt | I am considering moving my server system from Ubuntu to FreeBSD as I make some pretty massive network changes in the near future. I can always find out the hard way, though. :-) |
| 04:40.50 | frogonwheels | mbt: can't you just run a virtual box - or is there hardware involved? |
| 04:41.04 | frogonwheels | mbt: I mean to test it out. |
| 04:41.33 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 04:42.04 | aiksa[LV] | mbt - I have tried that (for the sake of truth that was a long long time ago), then it was far from ready for the production enviroment |
| 04:42.30 | mbt | I was planning on testing in a VM partially before putting it on the server, but I run Asterisk on the only public IP I have, which is currently an Ubuntu server. VM testing will be helpful, but likely not conclusive. |
| 04:42.50 | aiksa[LV] | but if you need a slimmer n*x* than that bloatware ubuntu, there are number of small and efficent distros out there |
| 04:42.59 | keebler | If anyone wants a "cheap" minimalistic Wireless Bridge that runs linux, I might have the toy for you. :) |
| 04:43.07 | keebler | erm Router/Bridge |
| 04:43.25 | mbt | LOL. I like Ubuntu for servers, it's not as bloated as you might think, configured properly. :) |
| 04:43.30 | keebler | Damn thing is only 3"x4". |
| 04:43.32 | frogonwheels | keebler: you still got that WRT kicking about :) |
| 04:43.40 | mbt | It's just that I have always liked FreeBSD on servers better. |
| 04:43.47 | keebler | frogonwheels: Yeah, but I'm playing with another toy now. |
| 04:43.55 | frogonwheels | oh? |
| 04:44.01 | keebler | mbt: High Five on the FBSD. |
| 04:44.06 | keebler | frogonwheels: Yeah.. I'll link ya. |
| 04:44.51 | aiksa[LV] | mbt, well I too prefer FreeBSD, but not for asterisk |
| 04:44.57 | keebler | frogonwheels: http://store.wisp-router.com/wri/itemdesc.asp?ic=EZ-Go-2&eq=&Tp= there ya go. I took the whole thing apart within 5 minutes of it arriving on my doorstep. :) |
| 04:45.19 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 04:45.19 | keebler | aiksa[LV]: Any disadvantages i should be aware of with FBSD+Asterisk? |
| 04:45.31 | keebler | aiksa[LV]: Cause thats what I'm running now. |
| 04:45.43 | keebler | aiksa[LV]: I'm genuinely curious/worried. |
| 04:46.23 | aiksa[LV] | keebler: As I said my experince is rather old by now, but I ran into several timing issues as the kernel modules for timing differes pretty much between these two |
| 04:47.16 | keebler | aiksa[LV]: Ah. hmm. I haven't gotten that deep into the Asterisk inner workings to notice any issues of timing. |
| 04:47.48 | aiksa[LV] | keebler: If I am not mistaken - benjk did some stuff to have it more or less reliabbly running on BSD and MacOS. |
| 04:48.09 | keebler | aiksa[LV]: Ah. That might explain why it works. :) |
| 04:48.29 | *** join/#asterisk stanthemancan (n=stan_man@S010600195b3059b4.gv.shawcable.net) |
| 04:48.34 | DJ_HaMsTa | woot i got asterisk registered with les.net, how do i configure my sipura phone to asterisk ? |
| 04:48.38 | stanthemancan | Hey, need help with the Asterisk Gui if possible... |
| 04:48.47 | aiksa[LV] | but i am not sure if that was included in any offical asterisk releases or existed just as a patch set. |
| 04:49.09 | DJ_HaMsTa | #asterisk-gui |
| 04:49.37 | keebler | frogonwheels: That "wireless bridge" I linked you can run off of 5.5VDC/500mA. I was looking for a tiny embedded CPE that could rival the WRT54G. Supposedly this model, with its 20dBi ant can get 3 miles. |
| 04:50.18 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 04:56.57 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:04.17 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 05:11.59 | LemensTS | $asm->Originate($channel, '', '', '', 'DeadAGI', 'test.php', '', '', 'var1=41283'); now in test.php, wouldnt i catch 41283 simply by calling $var1 |
| 05:18.17 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 05:19.00 | [TK]D-Fender | LemensTS: No. |
| 05:21.16 | LemensTS | Ive tried just setting it to '41283' and calling $argv, $argv[1] with no luck. |
| 05:21.41 | *** join/#asterisk neurosys (n=vinix@c-71-196-8-216.hsd1.fl.comcast.net) |
| 05:28.28 | sipy | TK: don't you ever sleep? |
| 05:29.08 | sipy | got any pointers on tracing down mwi not lighting? |
| 05:30.09 | aiksa[LV] | LemensTS: I think that variable was passed not to PHP but rather to channel in ast |
| 05:31.40 | LemensTS | aiksa: yea im trying $id = $agi->get_variable(var1); now |
| 05:31.52 | bmoraca | sipy: what phone model and asterisk version? |
| 05:36.27 | [TK]D-Fender | sipy: I try to set aside 7-8 minutes a night where I can... |
| 05:36.38 | aiksa[LV] | [TK]D-Fender: monster ... |
| 05:36.49 | sipy | polycom 501, I'm on to something. xml files I'll try |
| 05:37.15 | aiksa[LV] | my wife would have deleted the records in our house-book years ago if I were on a schedule like this |
| 05:37.17 | sipy | 7 8 minutes? I need twice that in hours |
| 05:37.25 | [TK]D-Fender | sipy: "mailbox=123@contextinvoicemail.conf" <- sip.conf peer entry |
| 05:38.02 | sipy | ah yeah, I forgot about that |
| 05:38.07 | [TK]D-Fender | sipy: You don't need anything set on the phone |
| 05:38.21 | sipy | ok I'll try that thanks |
| 05:38.22 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 05:39.03 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58-8-2-35.revip2.asianet.co.th) |
| 05:40.30 | HaMYaI | Hi, one of my users is trying to register using an incorrect sip password. Is there a way to see that password or to allow him to register using any password? |
| 05:40.52 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-c17b4aa7d8dc7f2d) |
| 05:41.07 | [TK]D-Fender | HaMYaI: No, and no. |
| 05:42.21 | HaMYaI | [TK]D-Fender: any suggestion? Unfortunately, I have no other way to communicate with him |
| 05:42.40 | bmoraca | tell him not to fuck with his phone next time... |
| 05:42.59 | [TK]D-Fender | HaMYaI: He has internet acess and you have no means of communication? |
| 05:43.10 | [TK]D-Fender | .......... |
| 05:43.18 | HaMYaI | bmoraca: that way I still need some types of communication with him |
| 05:43.34 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:44.14 | HaMYaI | [TK]D-Fender: hmm, that user isn't even have a PC, only a sip phone |
| 05:44.45 | [TK]D-Fender | HaMYaI: No normal telephone there? Cell? MAIL? |
| 05:44.50 | HaMYaI | and he's behind firewall |
| 05:45.17 | harry_v | exit |
| 05:45.20 | HaMYaI | [TK]D-Fender: trying to find his phone number actually |
| 05:45.57 | sipy | 192.168.1.2 |
| 05:46.17 | bmoraca | 172.168.1.1 |
| 05:46.30 | [TK]D-Fender | 8.6.7.5.3.0.9? |
| 05:47.58 | drmessano | 312.67.43.245 <-- IP 5,6, CSI HAZ IT |
| 05:48.12 | [TK]D-Fender | drmessano: I remember that episode :) |
| 05:48.18 | [TK]D-Fender | drmessano: I laughed instantly :) |
| 05:48.25 | drmessano | Hell yes |
| 05:53.32 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 06:00.40 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 06:02.27 | *** join/#asterisk jocko (i=daemon@c-98-203-142-135.hsd1.wa.comcast.net) |
| 06:04.44 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 06:07.37 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 06:08.54 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 06:13.20 | drmessano^ | hmmm |
| 06:15.27 | *** join/#asterisk jocko (i=daemon@c-98-203-142-135.hsd1.wa.comcast.net) |
| 06:15.44 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 06:16.27 | [TK]D-Fender | Ok... I'm done for the night... later all |
| 06:16.32 | *** join/#asterisk ScribbleJ (n=sj@c-67-172-6-141.hsd1.il.comcast.net) |
| 06:16.43 | jocko | Hello, Anyone in here have experience putting linux/asterisk on a flash card? |
| 06:16.55 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 06:17.26 | ScribbleJ | jocko, not with asterisk, but when I want linux on flash, I usually just grab the Ubuntu install cd, boot itup, there's anoption int he admin menu to install to usb stick, work wonders; theny ou'd just aptitude install asterisk in there... you could go simpler of course. |
| 06:18.07 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 06:19.19 | jocko | I was looking to get a bare linux install, asterisk and freepbx on a 1gb flash card and have it load into ram on boot. |
| 06:20.16 | jocko | I'm not looking to reinvent the wheel if someone has already done this. |
| 06:21.09 | jocko | Xorcom's TS-1 was setup like this with 512mb flash and 512mb ram |
| 06:22.10 | *** join/#asterisk rhombus (n=rhombus@dsl-vlan435-66-18-218-36.nucleus.com) |
| 06:22.10 | jocko | unfortunately it uses asterisk 1.1 so I'm looking to use some more current |
| 06:22.40 | rhombus | Can you use the VMCOUNT() function to tell you how many *new* messages are in the specified mailbox? |
| 06:23.20 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 06:24.17 | rhombus | or is it just a matter of specifying the folder as 0? |
| 06:25.46 | *** part/#asterisk jocko (i=daemon@c-98-203-142-135.hsd1.wa.comcast.net) |
| 06:33.58 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ebcc1f712ccecd0d) |
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| 07:41.08 | StanManCan | How can i tell if asterisk is running? |
| 07:41.19 | StanManCan | I've installed the gui but can't access it through my browser |
| 07:43.08 | *** join/#asterisk baliktad (i=baliktad@c-24-16-23-12.hsd1.wa.comcast.net) |
| 07:43.13 | yang | StanManCan: "ps wuxa |grep asterisk" will tell you |
| 07:43.34 | StanManCan | er |
| 07:43.37 | StanManCan | i'll have to be back about that one |
| 07:43.42 | StanManCan | formatting for the fourth time |
| 07:43.46 | StanManCan | really wish i could get this running ... |
| 07:43.53 | StanManCan | :( |
| 07:45.00 | StanManCan | is there a preferred OS for asterisk ? |
| 07:45.48 | *** join/#asterisk noneo (n=ankamins@gate.gratka.pl) |
| 07:48.03 | drmessano | *nix |
| 07:48.10 | GameGamer43 | StanManCan: what os are you running? |
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| 07:56.57 | brunner | ROFL! Look what I just found: http://www.flickr.com/photos/chrisbrunner/1042239/ |
| 07:58.12 | drmessano | That makes me not want to use Asterisk anymore |
| 07:59.20 | *** part/#asterisk noneo (n=ankamins@gate.gratka.pl) |
| 08:00.17 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:00.30 | dominic1 | why am I not getting this http://www.voip-info.org/wiki/view/asterisk+manager+events#ReloadEvent event with my asterisk 1.6? |
| 08:04.26 | drmessano | That documents is five years old |
| 08:04.30 | drmessano | Maybe somethings changed |
| 08:04.48 | dominic1 | in 1.4 it was no problem |
| 08:04.56 | dominic1 | :-( |
| 08:05.56 | drmessano | Sorry |
| 08:06.01 | drmessano | I would demand a refund |
| 08:06.03 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 08:06.14 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:07.41 | ChannelZ | Anyone have a source for milliwatt test numbers in CO (USA)? I've been hunting around but can't find any.. asked the phone installer today, and he first asked "why do you want that?" and after I explained why (to calibrate my TDM card) he said "We don't use those any more, I haven't had those numbers for years." |
| 08:07.57 | ChannelZ | I kinda got the impression he was lying |
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| 08:36.21 | mort_gib | join #citrix# |
| 08:36.32 | stintel | no way ! ツ |
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| 08:39.09 | mort_gib | stintel: Sorry, Still not fully awake! |
| 08:39.29 | stintel | mort_gib: get some coffee ;) |
| 08:39.46 | mort_gib | It's brewing as we speak :-) |
| 08:40.01 | stintel | ;) |
| 08:42.06 | mort_gib | Gunshot Expresso B-) |
| 08:42.36 | stintel | that'll wake you up ;) |
| 08:45.31 | *** part/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
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| 09:45.38 | rima | I have some basic questions about asterisk that i was hoping someone here could help me with, here goes... I'm upgrading my version of asterisk from a very old version, if I upgrade to 1.4 or 1.6 do I have to rewrite my dialplan in AEL or can i still use my old extensions.conf dialplan? |
| 09:49.15 | lanning | you can use the old extensions.conf, but you really have to do research on which apps you use and some syntax. there is an upgrade.txt file that points some gotchas. |
| 09:51.11 | rima | Ok, thank you lanning. |
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| 11:15.45 | dominic1 | again a little proble, |
| 11:16.36 | dominic1 | problem |
| 11:18.10 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 11:18.21 | dominic1 | if I create a conference with asterisk 1.6, in the conference asterisk sometimes cut the end of my sentences |
| 11:19.02 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
| 11:19.10 | ludan | hi |
| 11:20.16 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:20.44 | fiddur | dominic1: tweak the settings in meetme.conf... audiobuffers perhaps |
| 11:23.11 | ludan | how can I check if my fwd provider is reachable? |
| 11:23.23 | ludan | I had a configuration for which it was possible to call from a landline |
| 11:23.32 | ludan | and being fwd to the conference room |
| 11:23.50 | ludan | now I can get in the conf through SIP sw like ekiga |
| 11:23.58 | ludan | but not through landline anymore |
| 11:23.59 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 11:24.30 | Faustov | hi, does anyone have experience in configuring a GSM pci card with *? |
| 11:25.33 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.156) |
| 11:27.16 | dominic1 | thank you, audiobuffer helped |
| 11:28.11 | dominic1 | but it seems asterisk now always disables the mic of my conferencepartners when idle |
| 11:28.17 | dominic1 | is that normal behaviour |
| 11:29.02 | ludan | sen /etc/asterisk 83 # asterisk -rx "iax2 show registry" |
| 11:29.02 | ludan | Host dnsmgr Username Perceived Refresh State |
| 11:29.02 | ludan | 64.34.95.41:4569 N 760164 <Unregistered> 60 Timeout |
| 11:29.09 | ludan | I don't understand why |
| 11:29.11 | ludan | any clue? |
| 11:29.21 | ludan | it is with iax2.fwd.net.net |
| 11:30.23 | ludan | fwd-gw/760164 64.34.95.41 (S) 255.255.255.255 4569 UNREACHABLE |
| 11:30.25 | ludan | grrrr |
| 11:33.32 | Faustov | is celliax included in asterisk? |
| 11:34.34 | ludan | sorry what is celliax? |
| 11:35.25 | Faustov | Celliax is a GPL channel driver for Asterisk, chan_celliax, development and download site http://www.celliax.org |
| 11:35.40 | Faustov | http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network |
| 11:35.42 | ludan | how can I check if it is installad? |
| 11:35.58 | Faustov | nevermind, i browsed through available modules, it's not there |
| 11:36.32 | ludan | the funny thing is that last May this installation was working |
| 11:36.40 | ludan | I cannot understand what's going on right now |
| 11:38.11 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 11:41.07 | ludan | there should be a way to check it :( |
| 11:42.11 | *** join/#asterisk contrabanda (i=contraba@79.99.248.6) |
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| 11:44.44 | contrabanda | hello |
| 11:44.52 | contrabanda | i need help |
| 11:45.23 | contrabanda | i have E1 card |
| 11:45.32 | contrabanda | and connected to PSTN |
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| 11:52.45 | ludan | asen*CLI> iax2 show registry |
| 11:52.46 | ludan | Host dnsmgr Username Perceived Refresh State |
| 11:52.46 | ludan | 64.34.95.41:4569 N 760164 <Unregistered> 60 Timeout |
| 11:52.49 | ludan | this is the story |
| 11:53.01 | ludan | does not connect anymore to that damn server :( |
| 12:00.38 | *** join/#asterisk contrabanda (n=contr@92.241.69.134) |
| 12:01.57 | contrabanda | i have errors NOTICE[5037]: chan_dahdi.c:8704 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 12:02.02 | contrabanda | how can i fix it? |
| 12:07.32 | contrabanda | <PROTECTED> |
| 12:10.01 | *** part/#asterisk rima (n=chatzill@cpc2-harg2-0-0-cust648.leed.cable.ntl.com) |
| 12:11.44 | contrabanda | hellooooooo |
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| 12:45.36 | loompek | hi... |
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| 13:00.24 | yang | loompek: hi ! |
| 13:00.40 | loompek | would it be possible for asterisk to send sip command move temporary in case of unavailable users? |
| 13:01.03 | *** part/#asterisk n3glv (n=n3glv@c-71-60-107-110.hsd1.pa.comcast.net) |
| 13:03.21 | AdvoWork | anyone here use trixbox? |
| 13:04.07 | beek | AdvoWork: #trixbox |
| 13:05.05 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:06.19 | AdvoWork | beek, no one responding :S |
| 13:06.33 | beek | ~trixbox |
| 13:06.34 | jbot | [trixbox] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 13:06.46 | Mr_BOnD_007 | one question i want to ask |
| 13:07.49 | Mr_BOnD_007 | i am new @ asterisk i have 2 asterisk server configured with VICIDIAL i am adding agents in that so if i want to know that what is the configuration of asterisk how can i know ? |
| 13:09.15 | beek | AdvoWork: I used to use trixbox -- it was my first attempt at asterisk. It didn't take long to realize that it was more of a PITA to figure out how Trixbox did something so that I could modify it than it was simply to install Asterisk proper and learn to configure that. |
| 13:11.22 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 13:11.29 | AdvoWork | beek, thats the thing, im looking into cdr reports, and i want to know where trixbox inserts the data into the mysql cdr table, is there a function that actually gets the src/desintation and so on information(well there must be in order to be able to insert)? |
| 13:12.35 | beek | Asterisk does CDR under the covers. I don't know if trix does anything else. |
| 13:13.00 | AdvoWork | what do you mean? excuse my ignorance :p |
| 13:13.58 | beek | What I mean is that Asterisk does CDR reporting without there being anything in the dialplan to do it. You can add your own information at times, but for the most part it's on autopilot. |
| 13:14.29 | *** join/#asterisk DarkRift (n=dark@65.92.251.12) |
| 13:14.48 | beek | If you pull the source down you can find all of the CDR-related code in the cdr directory. |
| 13:16.36 | *** join/#asterisk Jeff_Phillips (n=ceramics@66-112-49-13.stat.centurytel.net) |
| 13:18.49 | *** join/#asterisk Mark17 (n=mark@freenode/sponsor/mark17) |
| 13:20.09 | Mark17 | hello, currently i set the caller id for a call in the sip.conf at the section for that sip account, but is it possible to have a different internal caller id and an other extrenal caller id? |
| 13:20.38 | Mark17 | for internal use the short sip trunk number would be nice, but for external use a normal phone number should be displayed |
| 13:21.52 | beek | Mark17: Set your callerID information in your dialplan before dialing the call. |
| 13:22.26 | AdvoWork | beek, yeah ive got the source files, i can get the data fine by doing a query, ie select src from cdr; that gives me the src information BUT only once ive ended the call |
| 13:22.47 | AdvoWork | it must be End Call > Insert Data so basically the cdr report isnt realtime(but close) |
| 13:23.01 | AdvoWork | im trying to see where it gets inserted, or what function it uses to get that information |
| 13:23.26 | AdvoWork | ie trixbox/asterisk must have a function controlling an incomming call, containing information, and from there passes it to the reporting tools |
| 13:23.31 | AdvoWork | so im trying to find that :S |
| 13:26.02 | Mark17 | beek: is it possible to make a dialplan per sip account? |
| 13:26.26 | Mark17 | because the server is used by multiple companies for outgoing calls and this is needed for just 1 company |
| 13:26.38 | Mark17 | every company has an other external did |
| 13:27.03 | beek | Mark17: That's what the context parameter is for in SIP.conf |
| 13:27.48 | beek | AdvoWork: What are you trying to accomplish? |
| 13:28.15 | AdvoWork | i want to be able to write this information to my own things, there must be something though? |
| 13:28.35 | beek | There's AMI, if you really want realtime information. |
| 13:29.35 | elred | Hello there. I wrote a little AGI's script in python, it exit using sys.exit(0). But no wonder if I sys.exit(1) or -1, on the console it always appears like "AGI Script /tmp/appelsortant.py completed, returning 0". It's right because I exited with 0 return code, but it anyway print "returning 0" no wonder what was the return value of exit(). And, then, when I do a Verbose(${AGISTATUS}) just after calling the AGI in my dialplan, it always print "failure", e |
| 13:29.41 | elred | any idee why ? thanks |
| 13:29.58 | AdvoWork | beek, whats AMI? |
| 13:30.27 | beek | Asterisk Management Interface (configured in manager.conf). It's what makes your FOP work. |
| 13:30.48 | Mr_BOnD_007 | VICIDIAL ? what's this ? |
| 13:31.26 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:31.56 | AdvoWork | beek, but how is that realtime info? or how do you get that? |
| 13:32.11 | beek | AdvoWork: Read the book. There's a chapter on AMI |
| 13:32.14 | beek | ~thebook |
| 13:32.15 | 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 |
| 13:33.11 | beek | AdvoWork: short story: You can get every event that Asterisk is creating sent via AMI to a program of your design. |
| 13:34.19 | Mr_BOnD_007 | ty jbot i have allready downloaded that |
| 13:34.42 | *** join/#asterisk neurosys (n=vinix@c-66-176-24-16.hsd1.fl.comcast.net) |
| 13:34.46 | AdvoWork | beek, lookin now, so in theory, if u rang me, id be able to handle that src number straight away? |
| 13:35.30 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:37.14 | AdvoWork | beek, but it must already be enabled because the cdr reports get that information?unless they use other means? |
| 13:37.21 | kaldemar | AdvoWork: what are you trying to do? what do you mean by handle? |
| 13:38.58 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 13:39.00 | *** join/#asterisk LuisTorres (n=chatzill@a213-22-94-113.cpe.netcabo.pt) |
| 13:39.11 | rwaite | is there a way to define a channel variable for certain sip extensions? or a way to find out what extension is calling in the dialplan (instead of using the callerid) |
| 13:39.32 | AdvoWork | kaldemar, well in simple terms, say a call comes in, and it gets the destination,i want to know that destination as the call rings, realtime |
| 13:39.45 | AdvoWork | my ami is enabled and already set afaik |
| 13:39.58 | beek | AdvoWork: the easiest way to see this in action is to set youself up with a simple account in manager.conf and then telnet to port 5038 of that box, sign on and add 'events: yes', and watch the fun as you place a call. |
| 13:40.49 | kaldemar | depending on where and how you want to know it, you can also use the dialplan for many wonderful things. |
| 13:40.52 | contrabanda | hello |
| 13:40.56 | contrabanda | i have errors NOTICE[5037]: chan_dahdi.c:8704 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 13:41.00 | contrabanda | how can i fix it? |
| 13:41.59 | kaldemar | rwaite: setvar parameter in sip.conf or function SIPPEER |
| 13:42.05 | *** join/#asterisk path_ (n=path@215-127-21-190.adsl.terra.cl) |
| 13:42.38 | rwaite | awesome, thx |
| 13:42.40 | LuisTorres | Howdy |
| 13:43.09 | LuisTorres | Theres any way to detect fax machines? |
| 13:43.11 | LuisTorres | like AMD |
| 13:43.32 | AdvoWork | ive added a user and it says do "module reload manager" and it says: -bash: module: command not found |
| 13:44.15 | kaldemar | LuisTorres: parameter faxdetect |
| 13:44.17 | Mark17 | beek: with AMI it is possible to let it do something when a call is started (some is calling and is at the beginning of the dialplan), someone does pickup the phone and when the call is ended? |
| 13:44.41 | kaldemar | AdvoWork: do it in the asterisk cli, not bash |
| 13:44.46 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:44.49 | LuisTorres | Kaldemar: thanks mate |
| 13:45.11 | AdvoWork | kaldemar, how do you actually get to that? |
| 13:45.39 | kaldemar | AdvoWork: asterisk -r. this is the point where you should stop asking here and go read the book. |
| 13:46.22 | kaldemar | then come back when you have at least some kind of conception of how asterisk works. |
| 13:47.24 | kaldemar | or just do stuff the trixbox way, whatever that is. you're going to end up with a major headache anyway if you plan on modifying things manually and using a GUI in parallel. |
| 13:47.46 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 13:47.50 | Mr_BOnD_007 | rEAding Bk |
| 13:47.55 | AdvoWork | wel ive added the user as requested, trying to telnet in(ive set the ip to allow as it states) and states: telnet: could not resolve myip:5038/telnet: Name or service not known |
| 13:48.00 | AdvoWork | kaldemar, yeah ive ordered the book |
| 13:48.09 | AdvoWork | 2 days ago when i took this project on |
| 13:48.18 | AdvoWork | up until now ive not had much experience with it |
| 13:48.21 | Mr_BOnD_007 | AdvoWork http://downloads.oreilly.com/books/9780596510480.pdf download it |
| 13:48.24 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:49.37 | kaldemar | AdvoWork: by all means download the pdf version too to get your hands on it. |
| 13:49.43 | beek | AdvoWork: the way to telnet in is: telnet myip 5038 |
| 13:50.46 | beek | AdvoWork: If you're expecting a logon screen you'll not get that. This acts more like an SMTP connection. So download the PDF of the book and read the chapter on AMI. |
| 13:51.02 | AdvoWork | beek, ive printed it out and am following what it says |
| 13:51.35 | beek | AdvoWork: It think AMI may provide you what you are looking for. |
| 13:51.37 | AdvoWork | ie telent in, done that, type: Action: login Username: myuser Secret: mysec <ENTER> it just says: Response: Errorb Message: Missing action in request |
| 13:51.39 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 13:51.59 | beek | AdvoWork: You are doing that on separate lines, right? |
| 13:52.02 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:52.06 | beek | Action: login |
| 13:52.10 | beek | Username: myuser |
| 13:52.16 | beek | Secret: mysecret |
| 13:52.19 | beek | Events: yes |
| 13:52.24 | beek | <ENTER> |
| 13:52.53 | AdvoWork | yeah, im typing Action: login <enter> to get to next line, that not right then? |
| 13:52.58 | AdvoWork | only thing I didnt do was events |
| 13:53.05 | beek | AdvoWork: That is correct. |
| 13:53.15 | AdvoWork | letme try again |
| 13:53.26 | beek | AdvoWork: I think its: Action: Logon |
| 13:53.41 | dlewis | nice |
| 13:53.57 | AdvoWork | Response: Success |
| 13:53.57 | AdvoWork | Message: Authentication accepted |
| 13:54.13 | beek | AdvoWork: You're in. Now place a call and watch what you get. |
| 13:54.33 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:55.10 | AdvoWork | beek, yeah just tried that lol, useful! do you need to quit out of it in any particular way? Also asterisk -rvvvvv did similar, but i couldnt get that data.. so if im using php or similar, how am i still going to get that data? |
| 13:55.20 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 13:55.49 | beek | AdvoWork: To exit telnet hit "Ctrl-]" and that will get you back to a prompt, then 'quit' |
| 13:56.41 | beek | AdvoWork: To get that information you need to do some socket programming. I'm sure that PHP has a library to make that easy. Your program has to act like you typing, then receive and parse the data as it comes. |
| 13:56.59 | AdvoWork | ahh, could possibly do system calls? |
| 13:57.28 | Mark17 | AdvoWork: PHP has an option for it on the local commandline or with ssh (for ssh an additional package is required) |
| 13:57.54 | AdvoWork | Mark17, yeah I think ive got it working before |
| 13:58.37 | Mark17 | http://nl.php.net/exec << have a look at that documentation |
| 13:58.46 | Mark17 | for doing it on the local system |
| 13:59.26 | beek | AdvoWork: I don't do PHP programming, so I'm not sure what you'll need. But I think that you'll get the info you're looking for from AMI. |
| 13:59.33 | beek | AdvoWork: Have fun! |
| 13:59.45 | Mr_BOnD_007 | beek asterisk we need to do socket programming ? |
| 14:00.16 | Mark17 | is it possible to let asterisk do something when a call is started (some is calling and is at the beginning of the dialplan), someone does pickup the phone and when the call is ended? |
| 14:00.44 | beek | Mr_BOnD_007: I don't know about you, but AdvoWork will need to. |
| 14:01.03 | Mr_BOnD_007 | okie beek sory to disturb u |
| 14:01.05 | Mark17 | AdvoWork: if you want to connect to a tcp socket with php you should look at nl.php.net/fopen |
| 14:01.31 | AdvoWork | Mark17, just looking at that now |
| 14:02.27 | Mark17 | ok |
| 14:02.50 | AdvoWork | Mark17, got it working as such, but im doing://$process = proc_open('telnet ip5038', $descriptorspec, $pipes, NULL, NULL, $other); so can pass one line, but not all the login lines and so on? |
| 14:04.13 | Mark17 | there was an option, but i dont remember it to be honest (you could ask in ##php) |
| 14:07.16 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
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| 14:10.55 | elred | nobody know why my AGISTATUS is set to FAILURE even tho it exited with return code 0 ? |
| 14:11.54 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 14:12.06 | elred | oops sorry, it's working. |
| 14:12.10 | elred | ignore my message |
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| 14:24.01 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:24.08 | Katty | Wocka. |
| 14:24.28 | beek | morning jaytee |
| 14:24.31 | beek | morning [TK]D-Fender |
| 14:24.45 | [TK]D-Fender | beek: mornin' |
| 14:24.49 | [TK]D-Fender | Katty: Mew. |
| 14:24.57 | Katty | pamples [TK]D-Fender |
| 14:26.58 | jaytee | mornin beek |
| 14:27.07 | jaytee | mornin [TK]D-Fender |
| 14:27.19 | jaytee | morning Katty |
| 14:28.26 | Katty | hai jaytee! |
| 14:28.28 | Katty | hugs jaytee |
| 14:28.49 | jaytee | hugs Katty |
| 14:28.56 | jaytee | have ya thawed out yet down there? |
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| 14:31.29 | rwaite | !${DB(test/boolean)} |
| 14:31.39 | rwaite | if test/boolean is 1, that should return 0, right? |
| 14:32.25 | Katty | jaytee: trying. |
| 14:32.32 | Katty | jaytee: everything's still a bit crunchy and cold. |
| 14:33.11 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
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| 14:36.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:37.18 | Katty | hugs [intra]lanman |
| 14:38.10 | *** join/#asterisk MONSIEUR_CHEVAL (n=CHEVAL_@bebif01.ulb.ac.be) |
| 14:38.18 | MONSIEUR_CHEVAL | salut àtous |
| 14:38.21 | MONSIEUR_CHEVAL | ici CHEVAL |
| 14:38.56 | [intra]lanman | hugs Katty |
| 14:39.03 | [intra]lanman | Katty: hi howarya |
| 14:39.21 | [TK]D-Fender | MONSIEUR_CHEVAL: Va-t'ens mon ostie! :p |
| 14:39.44 | *** part/#asterisk MONSIEUR_CHEVAL (n=CHEVAL_@bebif01.ulb.ac.be) |
| 14:40.02 | [TK]D-Fender | lol |
| 14:40.31 | [TK]D-Fender | grants people a warm welcome...... kerosene included :D |
| 14:42.01 | [TK]D-Fender | Too bad he couldn't take a joke :) |
| 14:44.51 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:45.53 | sipy | up from your 8 minute nap I see TK? |
| 14:46.38 | sipy | francais TK? |
| 14:49.51 | [TK]D-Fender | sipy: ...... we've met before :) |
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| 14:50.55 | *** join/#asterisk ZefK (n=ZefK@wsc-fo.b.astral.ro) |
| 14:52.30 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 14:54.41 | ZefK | Hi. How can I monitor how many channels are used on an ISDN PRI span? thx. |
| 14:54.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:56.30 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 14:56.45 | [TK]D-Fender | ZefK: "show channels concise" , "zap show channels" , "core show function GROUP_COUNT" |
| 14:57.11 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 14:57.13 | rwaite | core do what i want? |
| 14:58.12 | *** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca) |
| 14:58.25 | [TK]D-Fender | rwaite: [09:31]<rwaite>if test/boolean is 1, that should return 0, right? <- nope |
| 14:59.09 | rwaite | i'm like --><-- this close to writing a wrapper around calling a perl script and doing all my logical decisions there |
| 14:59.49 | *** join/#asterisk mocker (i=ksexton@198.247.173.227) |
| 15:00.25 | AdvoWork | any reason why when im using the AMI its doing: Message: Authentication accepted Response: Error Message: Missing action in request |
| 15:01.00 | [TK]D-Fender | rwaite: have fun. Total overkill, but whatever |
| 15:01.12 | mocker | Having a problem w/ this Polycom SoundStation. Getting 'username mismatch, have <1224>, digest has <>' and it won't register. |
| 15:01.17 | [TK]D-Fender | AdvoWork: Maybe because you're missing an Action..... |
| 15:01.38 | rwaite | it's just frustrating and non-intuitive to me |
| 15:02.26 | *** join/#asterisk chendy (n=chatzill@58.60.30.104) |
| 15:03.13 | AdvoWork | im not though |
| 15:04.11 | mocker | And registration failed Username/auth name mismatch |
| 15:04.40 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 15:06.14 | mocker | Any ideas? |
| 15:06.39 | ZefK | [TK]D-Fender, thx. Is it possible to call a function from CLI or only from dialplan ? |
| 15:09.07 | Jeff_Phillips | I have a weird DTMF problem when trying to use in-call feature codes on the receviing extension |
| 15:09.17 | mocker | Ugh, that sucked. |
| 15:09.48 | mocker | "Third Party Name" was the option that I had to set for the Polycom. |
| 15:09.51 | mocker | Didn't even think that was useed. |
| 15:10.01 | Jeff_Phillips | Example, I call from 110 to 130. Answer 130. I press *2 to try to perform an attended transfer. The 130 extension hears the "transfer?" prompt -- so the system recognized the command. But the extension that placed the call hears the tone stuck as though it is pressed indefinately until the call hangs up |
| 15:10.35 | mocker | So for IRC logs: "Third Party Name" in Polycom Web Interface is what you need to put the username in for the "username mismatch" error |
| 15:12.01 | [TK]D-Fender | ZefK: you can fake it from CLI with some effort |
| 15:14.28 | jplank | can anyone tell me if this makes sense. Polycom IP550 phones (also happens on 330) first call comes in, shows up on the first line appearance, everything is fine. Second call comes in, as soon as the second line key starts flashing, the user can't hear the first caller anymore, after about 3 seconds of ringing, the first caller gets automatically put on hold, if the user picks up the second caller, they can't hear them, but the cal |
| 15:15.10 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 15:21.42 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:22.24 | rwaite | jplank: you need to setup two lines in the config |
| 15:22.33 | *** join/#asterisk p0y (i=p0y@124.104.117.45) |
| 15:22.42 | rwaite | even if they only have one "extension" you need to set up both lines to use that extension |
| 15:22.51 | jplank | they have 4 line keys already setup |
| 15:23.41 | rwaite | i dont know then, i have 330s here and that was my problem. |
| 15:26.32 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
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| 15:26.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:27.49 | mikealeonetti | when I or my company takes somebody off hold I can't hear anything but the customer seems to still remain on hold... I'm not sure how this started. Has anybody ever heard of it? |
| 15:27.50 | rwaite | http://pastebin.com/m5a5d88ea |
| 15:27.58 | rwaite | for example, that's for one of my 330s |
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| 15:46.53 | jplank | rwaite, it actually seems like a codec issue that I can't figure out |
| 15:47.01 | jplank | I just found out it doesn't happen on internal calls |
| 15:51.58 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 15:52.53 | deadpigeon | Hi. Just wondering, I've got a pri trunk, and I keep seeing B-Channel 1-23 successfully restarted on span 1 pretty often, a few times an hour. Is this typical behavior for the span to continiously restart or should I be looking into why this is happening? |
| 15:54.20 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:55.14 | [TK]D-Fender | deadpigeon: "priresetinterval=never" |
| 15:55.32 | deadpigeon | [TK]D-Fender: Much thanks. |
| 15:55.57 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 15:56.09 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 15:56.38 | jayrod422 | everytime a agent on my asterisk hangs up a call they received from a queue asterisk show them as being unavailable (aka logged off) any idea why? |
| 15:58.49 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:59.42 | jjshoe | anyone have thoughts on this? it just goes in an oscilating pattern, and happens the second I start up asterisk: http://pastebin.com/d4317f9b |
| 16:00.57 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:01.59 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:02.54 | [TK]D-Fender | jayrod422: You aren't showing us anything, so no. |
| 16:03.13 | jjshoe | ~istplist |
| 16:03.16 | jjshoe | ~istp-list |
| 16:03.35 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-41b266586567acd8) |
| 16:03.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:03.43 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 16:03.56 | [TK]D-Fender | ~itsplist-us |
| 16:03.56 | 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 |
| 16:05.45 | *** join/#asterisk tjfontaine (n=tjfontai@oftc/staff/tjfontaine) |
| 16:08.25 | jjshoe | I'm going to kick this thing in the nutts |
| 16:10.01 | jjshoe | ~nat |
| 16:10.02 | jbot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:10.30 | [TK]D-Fender | ~sipnat |
| 16:10.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:10.38 | *** join/#asterisk n3hxs (n=HAMming@71.39.159.200) |
| 16:11.32 | sipy | ~nut |
| 16:11.48 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:11.53 | sipy | ~sipnut |
| 16:16.07 | seanbright | ~slapchop |
| 16:17.59 | jameswf | Anyone have friends near Baltimore |
| 16:18.16 | jameswf | *asterisk friends |
| 16:19.18 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:19.26 | mikealeonetti | what could be causing the phones to not be able to pick somebody up who is on hold? |
| 16:19.30 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 16:21.29 | rue_work | if they dont like being held and hung up |
| 16:21.31 | rue_work | ? |
| 16:22.22 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 16:23.27 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 16:23.59 | ChannelZ | Anyone have a source for milliwatt test numbers in CO (USA)? I've been hunting around but can't find any.. asked the phone installer today, and he first asked "why do you want that?" and after I explained why (to calibrate my TDM card) he said "We don't use those any more, I haven't had those numbers for years." |
| 16:26.31 | jameswf | ChannelZ: http://tinyurl.com/dlpjpt |
| 16:27.11 | mort_gib | <mikealeonetti> What phones?? |
| 16:27.30 | ChannelZ | Yeah. And most of those are from 4 years ago and don't work. |
| 16:27.31 | mikealeonetti | mort_gib: Cisco 7960 configured for SIP |
| 16:27.48 | mort_gib | OK, I had an isue with Snoms running 7.3.10a |
| 16:27.54 | mocker | ChannelZ: Playback the milliwatt sound file. :) |
| 16:28.17 | mocker | Or was it an app..? |
| 16:28.18 | path_ | :-) |
| 16:28.19 | mocker | can't remember now. |
| 16:28.30 | mocker | cmd_milliwatt |
| 16:29.49 | ChannelZ | it's an app.. but somewhat of a false test since if I call myself both the transmit and receive gains apply which might both be wrong |
| 16:29.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:30.27 | mikealeonetti | mort_gib: strangely enough, I can connect my phone outside of their network from here and put people on hold just fine with this phone and a Linksys phon |
| 16:30.27 | mikealeonetti | e |
| 16:30.46 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f82920213be7fb29) |
| 16:30.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:30.48 | jameswf | ChannelZ: how about a simple fxotune -i |
| 16:30.57 | mort_gib | Wow, you will want to have a close look at your firewall! |
| 16:32.02 | rue_work | I'm still looking for someone with enough aastra and or polycom experiance to help me get dialed digits to automatically dial, the phones dialplan is xx |
| 16:32.06 | rue_work | and I'v applied it |
| 16:32.10 | ChannelZ | fxotune is effective if your gains are proper which is the whole point of trying to find a reference milliwatt tone from the phone company |
| 16:32.16 | mikealeonetti | mort_gib: all of the phone that are having problems are on the same network and there are no firewall rules |
| 16:32.57 | mort_gib | So, they can't pickup parked calls?? |
| 16:33.26 | jameswf | ChannelZ: what makes you think your gains are improper, what is the issue your trying to fix? |
| 16:40.10 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 16:40.56 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:41.50 | *** join/#asterisk The_Boy_Wonder (n=The_Boy_@nat/digium/x-6d98b890d2cca6ce) |
| 16:42.17 | Katty | sighs |
| 16:42.22 | rue_mohr | reccommendations on ways of doing speed dialing? |
| 16:42.42 | *** part/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 16:42.42 | watchy2 | hugs katty |
| 16:42.44 | watchy2 | dont cry |
| 16:42.52 | Katty | so. i have a call to do, right? and i ask which vehicle they want me to take. they tell me to take the old beater truck because there are no other vehicles available. |
| 16:43.10 | Katty | I get halfway to my call and it DIES in the middle of a 4 lane highway |
| 16:43.14 | Katty | just straight up DIES. |
| 16:43.16 | watchy2 | a co i used to work for made me drive a vehicle called "oiler" |
| 16:43.33 | watchy2 | it was a giant full sized van |
| 16:43.34 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 16:43.40 | watchy2 | you work for a services company katty? |
| 16:43.46 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 16:43.47 | Katty | i guess. |
| 16:43.57 | Katty | stuff breaks, i go fix it. |
| 16:44.02 | watchy2 | what kinda stuff? |
| 16:44.02 | Katty | is that a services company? |
| 16:44.10 | watchy2 | if its for other companies, then yes |
| 16:44.14 | Katty | then ya |
| 16:44.25 | watchy2 | me to |
| 16:44.40 | watchy2 | but i'm looking at taking a Director of MIS job in little rock |
| 16:44.42 | Katty | there is ice all over the roads. |
| 16:44.45 | Katty | what were they thinking? |
| 16:44.46 | watchy2 | im tired of teh services bs |
| 16:44.53 | *** join/#asterisk kc2tnk (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:44.57 | watchy2 | i hate dealing with customers |
| 16:45.23 | Katty | gee, let's send someone in a truck, which dies all the time, up a 4 lane highway--and not even tell her the truck dies because we wouldn't want her to refuse to drive it! |
| 16:45.52 | watchy2 | haha |
| 16:46.16 | rue_mohr | Katty, you get paid by the hour, right? |
| 16:46.21 | Katty | did i mention there is a HUGE crack in the windshield? it won't pass inspection |
| 16:46.31 | Katty | rue_mohr: i get paid peanuts by the hour, yeah |
| 16:46.37 | Katty | rue_mohr: i am just above poverty line |
| 16:46.44 | rue_mohr | Katty, and you have a cellphone, right? |
| 16:46.58 | Katty | i prefer tincan and strings. |
| 16:49.07 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 16:49.59 | watchy | maybe they abuse you call your a girl |
| 16:50.18 | Katty | no, they take advantage of everyone here. it's not just me. |
| 16:50.40 | watchy | well, so does my company thats why i'm leaving and gonna let them tank |
| 16:51.00 | watchy | im the only voip dude here, and i got many implementations out there now |
| 16:51.11 | Katty | ya i'm the only voip person here |
| 16:51.17 | *** join/#asterisk fogo (n=Paul@69.169.132.35) |
| 16:51.17 | Katty | i'm also the only server person here |
| 16:51.24 | Katty | microsoft servers, specifically |
| 16:51.29 | watchy | i do it all |
| 16:51.30 | Katty | but whatever. |
| 16:51.38 | mort_gib | Katty: MS Servers :-( |
| 16:51.41 | Katty | i'm happy to have a job right now. |
| 16:51.43 | watchy | voip/web coding/microsoft shit etc |
| 16:51.51 | watchy | you a mcse katty? |
| 16:51.58 | Katty | mcp |
| 16:52.04 | watchy | ah how old are you? |
| 16:52.09 | Katty | ^_- |
| 16:52.17 | watchy | i got my mcse at like 20 |
| 16:52.18 | Katty | is sex and location coming next? |
| 16:52.23 | watchy | no |
| 16:52.25 | Katty | scowls |
| 16:52.27 | Katty | i'm 24 |
| 16:52.27 | watchy | but phone number is |
| 16:52.37 | sipy | haha |
| 16:52.40 | Katty | my phone number is 1800pissoff |
| 16:52.54 | beek | nice one! |
| 16:53.03 | Katty | ty. |
| 16:53.03 | watchy | i hate services work. its so annoying, not enough money in it either |
| 16:53.10 | beek | makes note for future reference |
| 16:53.12 | watchy | atleast in my area |
| 16:53.12 | Katty | not many jobs out here. |
| 16:53.21 | mort_gib | watchy: That depends |
| 16:53.24 | Katty | only 30k population |
| 16:53.26 | watchy | lots of jobs out there if you got skills |
| 16:53.29 | watchy | oh your town? |
| 16:53.34 | watchy | my town has 11k. |
| 16:53.35 | Katty | most of which is due to 2 huge hospitals |
| 16:53.46 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 16:54.09 | watchy | my towns big because of government bomb making contractors here |
| 16:54.14 | watchy | who i do alot of IT/voip work for |
| 16:54.23 | mikealeonetti | mort_gib: yes, any call placed on hold, when they pick it up they can't hear anything |
| 16:54.44 | fexy | free grand slams at Denny's today |
| 16:54.45 | mort_gib | Ok, and they are registered with your server?? |
| 16:54.48 | fexy | Until 2 |
| 16:54.55 | mikealeonetti | mort_gib: they are registered with their own server, |
| 16:54.55 | Katty | fexy: what?! |
| 16:54.56 | fexy | nationwide |
| 16:55.03 | mikealeonetti | mort_gib: in their local network |
| 16:55.07 | sipy | What nation? |
| 16:55.08 | fexy | I should have mentioned this yesterday haha |
| 16:55.10 | watchy | we dont have a dennys here |
| 16:55.12 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 16:55.16 | mort_gib | Not the server with the parked calls on?? |
| 16:55.19 | Katty | fexy: how do you know?! |
| 16:55.21 | watchy | we dont even have a ihop or waffle house |
| 16:55.25 | fexy | slickdeals |
| 16:55.26 | watchy | katty: the internet |
| 16:55.34 | Katty | woah |
| 16:55.36 | fexy | but it was on the super bowl commercial initially |
| 16:55.40 | Katty | denny's website says enjoy a free grand slam |
| 16:55.41 | watchy | u even have a dennys katty? |
| 16:55.46 | fexy | you get a free grandslam |
| 16:55.52 | fexy | and a coupon book |
| 16:55.54 | jaytee | if you go to babelfish.yahoo.com and set it to translate from Spanish to English and type in La Quinta it comes back in English as "Next to Denny's" |
| 16:56.13 | fexy | and if they make you wait too long you mike get a rain check |
| 16:56.21 | fexy | at least 500 rainchecks per dennys and 1000 coupon books |
| 16:56.21 | watchy | i guess they gonna be crowded as hell |
| 16:56.22 | Katty | woah |
| 16:56.25 | Katty | fexy's right |
| 16:56.30 | watchy | yea hes right |
| 16:56.36 | watchy | i also saw it on the superbowl |
| 16:56.37 | beek | What they lose in giving it away they'll make up in volume, right? |
| 16:56.39 | BCS-Satori | Which Digium telephony device would work best for a T1 Flex operating as a PRI to go into an HP DL320 running CentOS 5.2? |
| 16:56.55 | fexy | I'm going to get mine soon :D |
| 16:57.01 | Katty | woot |
| 16:57.03 | Katty | fexy: <3 |
| 16:57.03 | fexy | but I have a pack of smokes so I don't mind waiting :D |
| 16:57.04 | watchy | i wish i had a dennys |
| 16:57.08 | Katty | us girls are going to denny's for lunch |
| 16:57.09 | Katty | fexy: I LOVE YOU |
| 16:57.17 | fexy | haha :D |
| 16:57.24 | [TK]D-Fender | BCS-Satori: TE122P |
| 16:57.25 | watchy | wow a town of 30k has a dennys |
| 16:57.27 | watchy | i'm jealous |
| 16:57.39 | Katty | the thing about this town is weird. |
| 16:57.49 | Katty | we're the biggest 'city' in a whole crapload of farm area |
| 16:57.54 | watchy | haa |
| 16:57.57 | Katty | plus we have two huge hospitals |
| 16:57.59 | Katty | and a university |
| 16:58.01 | watchy | i live in arkansas |
| 16:58.09 | Katty | so the crowd that our city caters to, it's tourism |
| 16:58.15 | Katty | we have so many resturants |
| 16:58.16 | BCS-Satori | [TK]D-Fender: Thanks I see a TE122 on the website but not a "p" verison is that the same card |
| 16:58.26 | fexy | watchy no denny's in Arkansas? o_0 |
| 16:58.28 | [TK]D-Fender | BCS-Satori: same thing |
| 16:58.28 | Katty | olive garden, ocharlies, logans, apple bees, dexter bbq... |
| 16:58.36 | watchy | yes but 2 hours away |
| 16:58.44 | watchy | why would i pay $40 in gas for $4 free lunch |
| 16:58.46 | fexy | We have Red Robin |
| 16:58.57 | fexy | watchy for the experience? :p |
| 16:59.12 | watchy | i'm on a diet |
| 16:59.16 | watchy | down 87lbs so far |
| 16:59.19 | Katty | i'm on a seefood diet |
| 16:59.21 | fexy | sounds like you're making excuses now :p |
| 16:59.22 | Katty | see food, eat it |
| 16:59.40 | fexy | thinks watchy is afraid of Denny's |
| 17:00.06 | fexy | well be back later |
| 17:00.11 | fexy | to Denny's! |
| 17:00.14 | Katty | bai |
| 17:00.20 | Katty | wow, my day is suddenly brighter |
| 17:00.25 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 17:00.46 | mikealeonetti | Katty: you have an aura of misfortune |
| 17:01.14 | Katty | psh |
| 17:01.17 | Katty | free lunch notification |
| 17:01.24 | Katty | i'd hardly say that's misfortunate |
| 17:01.32 | BCS-Satori | [TK]D-Fender: Is there anything I should be concerned about or other devices for this product, we normally use a SIP Trunk and Audiocodes gateway. This will be our first internal card. |
| 17:01.36 | mikealeonetti | I smell death on you... |
| 17:01.48 | Katty | how does it smell? |
| 17:01.53 | mikealeonetti | walnuts |
| 17:01.58 | mikealeonetti | it smells of walnuts |
| 17:02.00 | Katty | hmm. i smell of walnuts. |
| 17:02.05 | Katty | neat. |
| 17:02.15 | watchy | you like in the south katty with lots of farm |
| 17:02.18 | watchy | ? |
| 17:02.30 | watchy | i don't live by any farms here in arkansas |
| 17:02.43 | watchy | i live by alot of rednecks though |
| 17:02.49 | Katty | yes, there is a lot of farmland around here |
| 17:02.55 | Katty | and lots of moo cows. |
| 17:03.00 | Katty | moo. |
| 17:03.11 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 17:03.16 | mikealeonetti | Katty: http://tshirtreviews.files.wordpress.com/2007/03/nuts120gallery_normal.jpg |
| 17:03.37 | Katty | mikealeonetti: :< |
| 17:03.45 | mikealeonetti | that's right |
| 17:03.53 | Katty | :< |
| 17:04.02 | mikealeonetti | be careful |
| 17:04.02 | watchy | i like moocows |
| 17:04.07 | Katty | k |
| 17:04.35 | [TK]D-Fender | BCS-Satori: personal recommendation Sangoma A101d |
| 17:06.13 | BCS-Satori | [TK]D-Fender: just curious any reason why? |
| 17:07.08 | tjfontaine | I have a legacy pbx that will send A B and D dtmf tones, I can use an application map to grab the D tone, but it seems that the A and B come too soon in the bridge for the map to pick them up anyone have any advice on catching the A and B tones? |
| 17:09.17 | *** join/#asterisk pythonist (n=paris@host170-225-dynamic.42-79-r.retail.telecomitalia.it) |
| 17:09.53 | pythonist | Hi, is there a way to wait for a phone call to finish? I can't figure out the correct application... |
| 17:12.47 | *** join/#asterisk n3hxs (n=HAMming@71.39.159.200) |
| 17:13.15 | watchy | i wish work wasnt so busy so i could work on this non bloated provision site for my polycoms |
| 17:16.44 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 17:17.03 | rue_mohr | oo tell me what you would do |
| 17:17.07 | watchy | anyone ever setup a card swipe on voip |
| 17:17.17 | rue_mohr | no |
| 17:17.28 | rue_mohr | hmm I dont have Labamba |
| 17:17.48 | Katty | la la bamba |
| 17:17.58 | rue_mohr | oops, did I say that... |
| 17:18.26 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 17:18.47 | [TK]D-Fender | BCS-Satori: Sangoma has always been no echo, no problems, period for 3 years straight |
| 17:18.50 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:19.01 | Katty | para bailar la bamba |
| 17:19.47 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-126-118.pools.arcor-ip.net) |
| 17:20.04 | Blackvel | is there any little php/cgi script to display Master.csv in a web gui? |
| 17:20.37 | Qwell | I'm sure there are dozens of things that can display a CSV file |
| 17:21.06 | Katty | cdr-stat is nice |
| 17:21.11 | Katty | it's php |
| 17:22.00 | Katty | jbot: asterisk-stat? |
| 17:22.02 | watchy | it wouldn't be very hard to make one for master.csv |
| 17:22.07 | Katty | REF: http://www.areski.net/asterisk-stat-v2/about.php |
| 17:22.11 | watchy | just use explode() on the file |
| 17:22.13 | *** join/#asterisk RouterWeasel (n=johnm@core.spokanecomputing.com) |
| 17:22.22 | Blackvel | most tools I find on voip-info or google are for mysql or database stuff. |
| 17:22.29 | Katty | yes. |
| 17:22.33 | Katty | and there is a reason for that. |
| 17:22.57 | Blackvel | I know...I dont have a very big environment which needs db |
| 17:23.14 | Blackvel | i'll checkout cdr-stat! |
| 17:23.16 | Katty | size matters not |
| 17:23.26 | [TK]D-Fender | Doesn't work... |
| 17:23.50 | [TK]D-Fender | [12:22]<Blackvel>most tools I find on voip-info or google are for mysql or database stuff. <- exactly |
| 17:23.59 | [TK]D-Fender | and nobody reads requirements before making suggestions. |
| 17:24.31 | Katty | that diet cherry coke at denny's is calling me |
| 17:24.36 | [TK]D-Fender | BCS-Satori: In case anyone was wondering why I mentioned a Digium card before a Sangoma one, one would have to read YOUR question. |
| 17:25.48 | *** join/#asterisk Khratos (n=khratos@190.166.103.180) |
| 17:26.19 | *** join/#asterisk adam000 (n=adam@c-76-97-76-93.hsd1.ga.comcast.net) |
| 17:27.42 | Khratos | Good $TIMEOFDAY |
| 17:28.01 | rue_mohr | [TK]D-Fender, I shoudl be able to set the phone up so if you dial a 2 digit extension it automatically sends? |
| 17:28.10 | *** join/#asterisk high-rez (n=gus@207-229-121-50.cortland.com) |
| 17:28.11 | rue_mohr | I set the dialplan to xx, but its not working |
| 17:28.32 | [TK]D-Fender | rue_mohr: CONTEXT |
| 17:28.46 | rue_mohr | polycom 601 |
| 17:28.56 | high-rez | I'm trying to slow down an FXO (dahdi device). It dials too soon after it takes the line off the hook and the first digit is getting missed by the remote switch.... Any suggestions on how to do this ? |
| 17:29.06 | [TK]D-Fender | rue_mohr: paste your XML line |
| 17:29.25 | [TK]D-Fender | high-rez: after the last "/" add "ww" |
| 17:29.38 | rue_mohr | <dialplan dialplan.applyToUserDial="1" dialplan.digitmap="xx" /> under <sip> |
| 17:29.40 | [TK]D-Fender | high-rez: eg : DAHDHI/ww1234567890 |
| 17:29.55 | [TK]D-Fender | rue_mohr: And whats in your phone files? |
| 17:30.00 | high-rez | Fender: Thanks man! |
| 17:30.15 | rue_mohr | nothing for dialplan |
| 17:30.15 | [TK]D-Fender | high-rez: eg : DAHDHI/g1/ww1234567890 |
| 17:30.35 | watchy | anyone ever do DNS server with SQL backend? |
| 17:30.49 | [TK]D-Fender | rue_mohr: thats really broken... that means they can't dial any normal looking # |
| 17:31.16 | rue_mohr | no they cant, because they would have to select which line it would go out on |
| 17:31.24 | high-rez | Fender: Yep... I did like this: exten => _1NXXNXXXXXX,2,Dial(DAHDI/G1/ww${EXTEN}) |
| 17:31.54 | [TK]D-Fender | high-rez: Should do. |
| 17:32.02 | high-rez | Fender: Thanks again man. :) |
| 17:32.22 | [TK]D-Fender | rue_mohr: And you've rbooted the phones? |
| 17:32.31 | rue_mohr | yup |
| 17:32.58 | rue_mohr | and if I just dial 12 it just sits there |
| 17:33.04 | [TK]D-Fender | rue_mohr: you seem to be missing several basic tags from that XML |
| 17:33.14 | [TK]D-Fender | rue_mohr: Go read your admin guide & samples again |
| 17:34.01 | *** join/#asterisk AJFisher (n=alex@82-70-11-70.dsl.in-addr.zen.co.uk) |
| 17:34.17 | rue_mohr | well I could make sure applytouserdial is set... |
| 17:34.24 | rue_mohr | no already done |
| 17:34.53 | AJFisher | Hi. I've just run into this bug. http://bugs.digium.com/view.php?id=14208 |
| 17:35.05 | AJFisher | It's not fixed in 1.6.0.5 |
| 17:36.33 | rue_mohr | timeout? |
| 17:38.17 | *** join/#asterisk riddlebox (n=user@75-105-81-181.cust.wildblue.net) |
| 17:39.12 | rue_mohr | [TK]D-Fender, is what I'm looking for part of the <sip> section? |
| 17:40.50 | *** join/#asterisk aatmaa_ (i=aatma@118.103.235.29) |
| 17:41.58 | rue_mohr | [TK]D-Fender, I think there is something fundamental I'm missing here |
| 17:42.34 | AJFisher | The comment in the bug report says the bug fix had been comitted in the 1.6 branch (which it has) and will make it into 1.6.0.4 and beyond (which it hasn't) |
| 17:43.09 | [TK]D-Fender | AJFisher: that is a topic for #asterisk-dev and you have provided no backup for your issue. |
| 17:43.44 | jjshoe | I'm spacing out, what's the tool you can use to check line voltage? |
| 17:43.50 | *** join/#asterisk Mattchis (n=IceChat7@c-98-199-191-45.hsd1.tx.comcast.net) |
| 17:43.57 | [TK]D-Fender | jjshoe: a multimeter |
| 17:44.24 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:44.46 | AJFisher | oh ok. I'll ask there. BTW, what do you mean by 'no backup for your issue'? |
| 17:44.58 | *** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-22d2530c0f5e6876) |
| 17:45.54 | [TK]D-Fender | AJFisher: "Hi, it doesn't work" - SHOW US |
| 17:46.37 | rue_mohr | well, he did say they didn't put the fix in the new versions, was it a patch that can be confirmed? |
| 17:47.21 | rue_mohr | [TK]D-Fender, is what I'm missing an applyTo ? |
| 17:48.12 | [TK]D-Fender | rue_mohr: go look at a stock config and TRY STUFF |
| 17:48.16 | AJFisher | it's a patch that's been applied. I experienced the exact same problem as is documented and been confirmed in the the URL I posted. Having found the bug report I was surprised to see that the version I'm running (1.6.0.5) was stilling affected. |
| 17:48.24 | [TK]D-Fender | rue_mohr: module unload chan_codependence.so |
| 17:48.25 | rue_mohr | I have... |
| 17:48.32 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733953.dsl.bell.ca) |
| 17:48.47 | AJFisher | The fix having been applied to the 1.6 branch before the release date of 1.6.0.5 ... |
| 17:49.07 | rue_mohr | [TK]D-Fender, but this is on the phone, so you dont ahve to press send... |
| 17:49.26 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 17:50.30 | AJFisher | On closer inspection I think I've solved the mystery. 1.6.0.5 wasn't always called 1.6.0.5. It started life as 1.6.0.3.1 and hence wouldn't have been made from the 1.6 branch but is presumely just 1.6.0.3 with a few cherry-picked fixes |
| 17:51.24 | Corydon76-dig | AJFisher: correct |
| 17:51.35 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:51.52 | rue_mohr | digitmap timeout? |
| 18:03.09 | *** join/#asterisk RouterWeasel (n=johnm@core.spokanecomputing.com) |
| 18:04.12 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 18:04.47 | *** join/#asterisk avb (n=avb@190.166.96.139) |
| 18:05.03 | avb | hey guys |
| 18:05.22 | rue_mohr | [TK]D-Fender, I cant understand whats wrong with a dialplan of xx that it wont take it |
| 18:05.45 | avb | can somebody help me with my problem. Im trying to fix an issue in a module in order to return BUSY in some cases |
| 18:06.06 | avb | how can i define BUSY return in code? |
| 18:06.13 | rue_mohr | sorry, have to work out some issues with my system first |
| 18:06.24 | avb | ast_softhangup() seems cant do this |
| 18:06.54 | avb | defining ast_chan->hangupcause is not a right way seems as well |
| 18:12.52 | LemensTS | Can you do sms in usa from asterisk to cell phones? From what I read this was not doable in usa... |
| 18:13.58 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 18:14.18 | icebrew54 | LemensTS: custom script? |
| 18:14.28 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 18:18.22 | Blackvel | have a good day/evening...bye |
| 18:21.19 | vader-- | Have any of you guys used a cisco ata 186 with a security panel? |
| 18:21.25 | vader-- | It dials but then the call drops |
| 18:21.28 | vader-- | eventually |
| 18:24.17 | *** join/#asterisk boomboom99 (n=boomboom@c-71-229-40-177.hsd1.ga.comcast.net) |
| 18:24.42 | fexy | I'm using sccp and I have a really basic dial plan and sccp.conf, but I can't call between phones. What am I miss? |
| 18:24.50 | fexy | err missing |
| 18:25.14 | fexy | Would a paste of my extensions.conf and a subset of my sccp.conf help? |
| 18:27.57 | boomboom99 | anyone know if it is possible to compile app_nv_faxdetect in Asterisk 1.6. If so, got a link to a howto? |
| 18:28.38 | coppice | doesn't 1.6 have SIP fax tone detect built in? |
| 18:28.46 | boomboom99 | really? |
| 18:29.18 | boomboom99 | I've got SIP to iaxmodem to Hylafax working (2 successful faxes received), but can't get it to detect |
| 18:29.37 | boomboom99 | I've been manually forwarding faxes to a fax extension that dials the iaxmodem |
| 18:34.48 | *** join/#asterisk baliktad (i=baliktad@c-24-16-23-12.hsd1.wa.comcast.net) |
| 18:37.52 | *** join/#asterisk Cubber (n=danky@static-74-41-185-190.br1.glv.ny.frontiernet.net) |
| 18:38.17 | Cubber | what version of asterisk is the most stable to use as a businesses sole phone system? |
| 18:40.06 | [TK]D-Fender | Cubber: typically the latest 1.4 series full release |
| 18:40.30 | Cubber | D-Fender: thanks so 1.6 is considered development then? |
| 18:40.58 | [TK]D-Fender | Cubber: Some might think a little too new... though so far pretty stable... but to start with I'd go with 1.4 |
| 18:41.34 | Cubber | D-Fender: great thanks for the input. |
| 18:43.44 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 18:45.45 | Corydon76-dig | [TK]D-Fender: it also depends on what he's using it for |
| 18:46.05 | Corydon76-dig | i.e. what features. There are a few areas I'd trust 1.6 more than 1.4 |
| 18:46.27 | Corydon76-dig | Queues, for one |
| 18:47.30 | *** join/#asterisk agx (n=badpengu@88-149-224-96.dynamic.ngi.it) |
| 18:48.05 | ScribbleJ | Whoot! |
| 18:48.05 | agx | is notifyringing=no in sip.conf to disable sending NOTIFY broadcasts when a telephone is ringing? |
| 18:48.25 | ScribbleJ | I have my speech recognition plugin working finally. |
| 18:49.04 | jaytee | ScribbleJ, a plugin? |
| 18:49.44 | [TK]D-Fender | Corydon76-dig: Will certainly note. What's the primary improvement VS 1.4? |
| 18:50.09 | Corydon76-dig | [TK]D-Fender: reference counting should improve stability |
| 18:51.00 | rob0 | Grrr, wctdm crashed my machine (which, incidentally, was not even running asterisk.) |
| 18:51.06 | jaytee | damn, I wish someone told me Asterisk was unstable when I was researching it last year instead of just now when I've got most of my users on it. |
| 18:51.32 | Corydon76-dig | unstable is relative |
| 18:51.46 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 18:51.52 | Qwell | Corydon76-dig: my relatives are unstable |
| 18:51.57 | ScribbleJ | jaytee, I basically am in the process of writing a drop-in replacement for LumenVox using the Asterisk Speech API, and various versions of Sphinx. |
| 18:52.14 | rob0 | TDM PCI Master abort, a gazillion times |
| 18:52.29 | ScribbleJ | If you had LumenVox and dialplan written for it, you could just drop this in and only change the line of code that tells which engine to use. |
| 18:52.31 | jaytee | I don't care if it's a redhead step-third cousin. I want stability. Stability NOW!!! |
| 18:52.51 | rob0 | Redheads have a certain charm. |
| 18:52.55 | russellb | jaytee: bugs.digium.com |
| 18:52.59 | jaytee | ScribbleJ, cool! |
| 18:53.29 | jaytee | rob0, I know. I'm a ginger kid myself |
| 18:53.31 | Corydon76-dig | russellb: I think he's being tongue-in-cheek |
| 18:53.56 | jaytee | Corydon76-dig, more like head up my own ass :-) |
| 18:54.13 | ScribbleJ | Jaytee, I'm pretty excited to release a /truly/ Free speech plugin to the community, even if it's worthless as anything other than reference. Although I gotta say I've been working on it in my spare time for a few weeks now and it seems to work better than I ever expected. |
| 18:54.14 | russellb | i see. |
| 18:54.26 | russellb | well so many people act like that and are serious, that it's hard for me to take the joke :-/ |
| 18:54.34 | Corydon76-dig | jaytee: don't you stick your tongue out at me, then |
| 18:54.49 | ScribbleJ | haaa |
| 18:54.55 | jaytee | actually I've got over 70 users with 74 by end of day on Asterisk 1.4.15 and it's pretty damn stable. |
| 18:55.02 | rob0 | Well I seriously DID have to hit the reset button to recover from my zaptel zap. |
| 18:55.25 | ScribbleJ | We just took our office off of Cisco Unity and moved to Asterisk -- stability is one thing Asterisk has over Unity in spades. |
| 18:55.33 | jaytee | russellb, you're brain is turning Vulcan from too much coding :-) |
| 18:55.34 | *** join/#asterisk Miccster (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 18:55.40 | russellb | jaytee: yes. |
| 18:55.40 | ScribbleJ | Actually, in my opinion, Asterisk's got everything over Unity; I hated that thing. |
| 18:55.57 | ScribbleJ | Asterisk over Unity.... hrm... something about that phrase is wrong. |
| 18:56.04 | vader-- | Have any of you guys used a cisco ata 186 with a security panel? |
| 18:56.23 | jaytee | russellb, I'd recommend going to Vegas and obtaining the services of a "physical therapist" |
| 18:56.24 | outtolunc | anyone see that commercial during superbowl.. 'don't be an asterisk' <G> |
| 18:56.53 | Corydon76-dig | vader--: my experience with security panels has led me to believe that they will generally not work at all with a PBX in the way |
| 18:57.05 | jaytee | I loved the one where the guy throws the snowglobe and hits his boss in the nuts. |
| 18:58.10 | vader-- | hmmm shitty |
| 18:58.13 | vader-- | the unit is dialing |
| 18:58.15 | vader-- | and connects |
| 18:58.17 | vader-- | just drops |
| 18:59.06 | rue_mohr | hmm I wonder if It will work if I do XX* |
| 18:59.08 | *** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 18:59.08 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 18:59.12 | *** part/#asterisk Mark17 (n=mark@freenode/sponsor/mark17) |
| 18:59.47 | boomboom99 | where can I research what fax detect features are available in 1.6? every search I do returns 1.4 and older :-( |
| 19:01.19 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 19:05.38 | [TK]D-Fender | boomboom99: the DOCS included with 1.6 |
| 19:11.36 | tAnkOSX | What ever happened to snap (anumber) ? |
| 19:11.53 | boomboom99 | found faxdetect=yes is possible in sip.conf, but: "This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection" |
| 19:12.03 | boomboom99 | grrr |
| 19:12.16 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 19:14.10 | *** join/#asterisk joesuffceren (n=chatzill@h125.219.135.98.ip.windstream.net) |
| 19:15.18 | joesuffceren | anyone know of a good (preferably but not necessarily free) 64 bit tapi driver for either asterisk or for cisco 7940 sip phones? |
| 19:16.35 | [TK]D-Fender | joesuffceren: Phones don't speak TAPI |
| 19:17.14 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 19:17.17 | Damin | Hey all.. |
| 19:17.40 | Damin | Anyone compiled asterisk-1.6.1-rc1 on Centos 5.2? |
| 19:18.00 | Damin | I'm running into the following error with func_curl |
| 19:18.01 | *** join/#asterisk jmworx (n=jeval@216.208.79.2) |
| 19:18.06 | joesuffceren | [TK]D-Fender: sorry for the confusion. I have a tapi driver for my snom 3xx phones, so that's where the confusion came in |
| 19:18.18 | Damin | <PROTECTED> |
| 19:18.29 | Damin | <PROTECTED> |
| 19:18.33 | joesuffceren | at any rate, I just want to be able to use TAPI applications (i.e. outllook and a ghastly old recruiting app) to be able to initiate calls |
| 19:18.51 | Damin | <PROTECTED> |
| 19:19.34 | tAnkOSX | As I already asked, what happened to Snap a number? |
| 19:19.44 | rue_mohr | flipflops between working on a speed dial system and getting the phones to use the dialplan he gave them |
| 19:19.47 | tAnkOSX | Wouldn't that be a solution for you joesuffceren |
| 19:19.59 | agx | boomboom99, until you have some megabit at a fix rate speed i wonder how you could receive more then 1 fax page over a SIP trunk |
| 19:20.19 | joesuffceren | tAnkOSX: sorry, didn't see the first one. I'm not familiar with snap a number. *googling* |
| 19:20.24 | tAnkOSX | :)))) |
| 19:20.29 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-187-233.lyn.east.verizon.net) |
| 19:20.33 | [TK]D-Fender | tAnkOSX: http://www.venturevoip.com/news.php |
| 19:20.44 | ScribbleJ | Waitasec, you can do FAX over SIP? |
| 19:21.00 | tAnkOSX | Thank you [TK]D-Fender |
| 19:22.08 | tAnkOSX | joesuffceren, i would check http://www.venturevoip.com/news.php?rssid=2099 |
| 19:25.59 | jmworx | Any asterisk "core developer" in here? |
| 19:26.21 | russellb_ | jmworx: #asterisk-dev is full of 'em ... depends on what you need, though ;-) |
| 19:26.26 | ScribbleJ | jmworx, #asterisk-dev |
| 19:26.30 | jmworx | Ah, sorry |
| 19:26.31 | ScribbleJ | gah |
| 19:27.00 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 19:27.03 | boomboom99 | yes, you can FAX over SIP. I've received 2 faxes (a 1 page and a 3 page) this morning. But, I've had to forward the call to a fax extension I have in my dial plan |
| 19:27.19 | boomboom99 | it calls an iaxmodem connected to hylafax |
| 19:27.26 | tAnkOSX | FAX and VOIP? I suggest you read http://www.soft-switch.org/foip.html |
| 19:27.26 | tAnkOSX | :) |
| 19:27.27 | rue_mohr | [TK]D-Fender, what do you think the best way to implement a speed dial is? |
| 19:27.37 | boomboom99 | just can't get automatic fax detection to work... |
| 19:27.55 | [TK]D-Fender | rue_mohr: feel free to get specific at any time |
| 19:28.07 | boomboom99 | read it... |
| 19:28.21 | agx | boomboom99, what do you use for fax detection? Over SIP you need an external app NVFaxDetect() remember you have to Answer() the channel for it to work, since it "listen" the incoming audio |
| 19:29.06 | agx | and IMHO if you have SIP just buy an additional number for faxes :) in Italy at least its free having an incoming number if you have credit onto the SIP account |
| 19:29.17 | rue_mohr | the receptionist has 32 numbers she commonly dials, right now one of the analog phones she has is equiped with 36 speed dial buttons, I am not going to buy an add on module for her, so I'm wondering what a good way of prodiding her with dial relief is |
| 19:29.49 | joesuffceren | tAnkOSX: I'll look into ADA. thanks |
| 19:29.59 | boomboom99 | agx, that's my original question: how to compile NVFaxDetect with Asterisk 1.6 |
| 19:30.05 | [TK]D-Fender | rue_mohr: You jsut said she has speed dial buttons. F-IN USE THEM :P |
| 19:30.25 | rue_mohr | and if your wondering, no I still cant get the polycom to automatically send when a user dials something that matches it |
| 19:30.54 | rue_mohr | [TK]D-Fender, we are removing her analog sets (all 4 of them) and replacing them with 1 polycom 601 |
| 19:30.57 | agx | boomboom99, i've a working version for 1.4 but not for 1.6 |
| 19:31.07 | rue_mohr | and all 4 of her call display boxes |
| 19:31.27 | [TK]D-Fender | rue_mohr: ok/fin/sure |
| 19:36.56 | Deeewayne | boomboom99: do you have libtiff installed ? |
| 19:37.27 | rue_mohr | wait, do the phones send when the dialed digits match or when they fail to be able to match it? |
| 19:38.08 | boomboom99 | Deeewayne, yes, libtiff was one of the deps for hylafax |
| 19:38.23 | rue_mohr | my understanding of the idea was that they send to the server when the dialed digits match the dialplan |
| 19:39.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:39.56 | *** join/#asterisk path_ (n=path@215-127-21-190.adsl.terra.cl) |
| 19:40.30 | *** join/#asterisk n3hxs (n=HAMming@71.39.159.200) |
| 19:41.10 | boomboom99 | to get fax over sip working, I loosely followed this howto: http://www.evaristesys.com/workshop/index.php/Inward_fax-to-email_gateway_with_Asterisk,_HylaFAX_&_IAXmodem |
| 19:41.14 | *** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it) |
| 19:41.18 | ruben23 | hi with sip.conf setting "qualify=1000" means...what |
| 19:43.12 | rue_mohr | ~pb |
| 19:43.12 | 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/ |
| 19:43.16 | ruben23 | :-(anyone...? |
| 19:43.18 | sipy | rue_mohr: if it doesn't match the dialplan on the phone, it never gets to asterisk |
| 19:43.47 | kaldemar | ruben23: sample config file will tell you that it makes asterisk qualify that the peer is reachable in 1000 ms. |
| 19:44.31 | rue_mohr | http://paste.debian.net/27583/ I dial 12 and the phone sits there looking dumb, it dosn't send the digits |
| 19:45.00 | sipy | it will send OPTIONS command with a an interval of 1000 |
| 19:46.13 | rue_mohr | sipy, it does match and the phone makes me press "SEND" which, is ofcourse, just a button with the text "SEND" stamped on it, it beinga polycom 601 |
| 19:46.14 | sipy | rue: what does the dial plan look like in the web interface? |
| 19:46.25 | rue_mohr | sipy, xx same as that config file |
| 19:46.37 | telnettech | what is the prefered script writing program that people working/using asterisk prefer? |
| 19:46.41 | rue_mohr | http://paste.debian.net/27583/ |
| 19:46.50 | [TK]D-Fender | Poeple configuring Polycom phones via the web interface should be dragged out and shot. Survivors should be shot AGAIN. |
| 19:47.15 | [TK]D-Fender | telnettech: What kind of "script"? |
| 19:47.15 | rue_mohr | [TK]D-Fender, but its a good way to confirm there isn't a local setting overriding your server config |
| 19:47.25 | boomboom99 | just broke down and bought another DID for dedicated faxing. $4.99/month thru Teliax. Not bad, but we only fax once every couple of months |
| 19:47.25 | [TK]D-Fender | rue_mohr: No, it isn't |
| 19:47.28 | sipy | just using it to verify what the phone has |
| 19:47.29 | agx | Is there a way to disable SIP NOTIFY Ringing for a single phone but only BUSY/UNAVAILABLE/IDLE ? I want only show when its busy and not when ringing (so the phone will not pickup it if the button is miss-pressed during a ringing phase). |
| 19:47.58 | telnettech | any scripts that need to collect info from the caller and either store or pull info from astDB or MySQL |
| 19:48.01 | *** part/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 19:48.10 | [TK]D-Fender | rue_mohr: You can't set the dialplan on the phone locally except for **VIA** the web interface, so if you ahve to go in there to see if you've been going in there then please just put the barrel to your head and pull the trigger now :) |
| 19:48.10 | telnettech | for dialplans |
| 19:48.28 | ruben23 | <PROTECTED> |
| 19:48.31 | [TK]D-Fender | telnettech: Who says you need anything externsal for this? |
| 19:48.41 | kaldemar | ruben23: no |
| 19:48.52 | [TK]D-Fender | telnettech: Much can be done entirely from the dialplan, and PHP is probably the most common language used for AGI |
| 19:49.28 | telnettech | ok thanks TK......im needing this so that I can plan training from company for me this coming year |
| 19:50.05 | rue_mohr | none the less, the dialplan on the phone (in its ram for freaking sake) is xx and if I dial 12 it does not send automatically, it makes me press another button (the one with the text "SEND" on it) |
| 19:50.46 | [TK]D-Fender | rue_mohr: not a complete description of your dial attempt... |
| 19:50.54 | rue_mohr | yes it is |
| 19:51.00 | [TK]D-Fender | rue_mohr: No, it ISN'T |
| 19:51.15 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:51.23 | *** join/#asterisk path_ (n=path@215-127-21-190.adsl.terra.cl) |
| 19:51.37 | rue_mohr | I aproach the phone press the "1" key, then i press the "2" key, I then expect it to send the digits automatically as the dialplan is xx but it dosn't |
| 19:51.55 | [TK]D-Fender | rue_mohr: it'll NEVER do it as described |
| 19:52.12 | rue_mohr | is it not possable to make it do this? |
| 19:52.21 | [TK]D-Fender | rue_mohr: how ELSE have you tried? |
| 19:52.24 | sipy | mine is working. |
| 19:52.37 | sipy | use the default as TK said and it works |
| 19:52.40 | [TK]D-Fender | sipy: there is something crucial in the what he is NOT saying |
| 19:52.49 | *** join/#asterisk beek_ (n=klinebl@pdpc/supporter/professional/beek) |
| 19:52.58 | rue_mohr | what? |
| 19:53.00 | [TK]D-Fender | sipy: there is something crucial in what he is NOT saying |
| 19:53.13 | rue_mohr | I do not press line keys or anyting, I press 1 and after that 2 |
| 19:53.21 | rue_mohr | thats it |
| 19:53.21 | [TK]D-Fender | rue_mohr: take the &^#$ing phone OFF-HOOK |
| 19:53.23 | rue_mohr | no set off hook |
| 19:53.30 | sipy | haha |
| 19:53.34 | [TK]D-Fender | rue_mohr: on-hook = no dialplan |
| 19:53.36 | rue_mohr | no, I expect it to use handsfree |
| 19:53.40 | rue_mohr | what!? |
| 19:53.44 | *** part/#asterisk jmworx (n=jeval@216.208.79.2) |
| 19:53.47 | rue_mohr | it wont auto handsfree? |
| 19:54.04 | [TK]D-Fender | rue_mohr: Expect? you EXPECT things? You wouldn't know the scientific process if it ran up and bit you in the face. |
| 19:54.11 | [TK]D-Fender | :0 |
| 19:54.12 | rue_mohr | !@#$#@!$ BUGGER! |
| 19:54.33 | rue_mohr | WHY dosn't it just go auto handsfree? |
| 19:54.46 | rue_mohr | bashes his head on the desk |
| 19:54.52 | [TK]D-Fender | rue_mohr: Because it isn't some 20 year old shit Nortel phone you're used to. |
| 19:55.00 | [TK]D-Fender | \/me HELPS |
| 19:55.02 | [TK]D-Fender | HELPS |
| 19:55.20 | [TK]D-Fender | WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham |
| 19:55.35 | rob0 | Wow. in 4 apparent seconds (seconds as determined by the system clock which hung, actually lasted about 7 hours), my TDM400 generated 300MB of syslog output, ended by the reset button. |
| 19:55.41 | [TK]D-Fender | pwned |
| 19:55.51 | rue_mohr | arg, so they have to either hit speakerphne manually or pick up the handset |
| 19:55.54 | [TK]D-Fender | rob0: WHEE! |
| 19:55.59 | [TK]D-Fender | rue_mohr: YES |
| 19:56.17 | [TK]D-Fender | rue_mohr: You're concept of thorough testing needs serious work. |
| 19:56.20 | rue_mohr | well, I think I can make them accept that |
| 19:56.26 | [TK]D-Fender | your* |
| 19:56.31 | rue_mohr | it shoudl do auto handsfree |
| 19:56.36 | rue_mohr | looks for a switch |
| 19:56.40 | [TK]D-Fender | rue_mohr: None. |
| 19:56.49 | tzafrir_laptop | rob0, what messages? |
| 19:57.31 | rob0 | TDM PCI Master abort, and portions thereof, and "last message repeated X times" |
| 19:57.42 | rue_mohr | you had me reading the dialplan section of the manual all day just to point out I have to take it off the hook for the dialplan to work |
| 19:57.46 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:58.01 | rue_mohr | so what is the blooming phone doing when I'm just dialing, entertaining me? |
| 19:58.10 | rob0 | google suggests an IRQ sharing problem, but this boot, it appears to be alone on IRQ 16. (x86_64) |
| 19:58.15 | [TK]D-Fender | rue_mohr: And I thought that for a second you'd have shown some brains and throroughly tested with the phone OFF-HOOK like the rest of the planet. |
| 19:58.56 | vader-- | hmmm i wonder if i should call cisco about this ATA 186 and trying to get it to work with this panel |
| 19:58.58 | rue_mohr | no I expect it to do "the user is dialing and were not offhook somehow, so I better get offhook, the handset is down, I'll use speakerphone" |
| 19:59.19 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 19:59.20 | [TK]D-Fender | rue_mohr: Did YOU head dial-tone while it was staying on-hook? NO. Please resume vigorous head-desking |
| 19:59.26 | [TK]D-Fender | hear* |
| 19:59.43 | [TK]D-Fender | rue_mohr: You've earned it |
| 19:59.50 | rue_mohr | hmm so I can also dial 14 and hit 'intercom' and it sends what it has |
| 20:00.29 | rue_mohr | might i point out there is no user manual for the 601? |
| 20:00.33 | sipy | thats funny |
| 20:00.41 | rue_mohr | I know how to program it, not use it |
| 20:00.52 | [TK]D-Fender | rue_mohr: PARDON?!?! |
| 20:00.56 | kaldemar | http://www.polycom.com/global/documents/support/user/products/voice/soundpoint_ip600_601_user_guide_sip2.0.pdf |
| 20:01.04 | kaldemar | rue_mohr: ^ what's that? |
| 20:01.07 | rue_mohr | ?? |
| 20:01.14 | [TK]D-Fender | reaches for his trusty ClueBat (tm) |
| 20:01.15 | rue_mohr | I onyl looked for 2 hours |
| 20:01.30 | ruben23 | <PROTECTED> |
| 20:01.49 | kaldemar | i managed to write polycom 601 manual into google under 2 seconds and get that as the first link. |
| 20:01.57 | [TK]D-Fender | rue_mohr: http://tinyurl.com/496svm |
| 20:02.11 | jaytee | the expression, "Couldn't find his own ass with both hands and a hunting dog" comes to mind |
| 20:02.15 | rhombus | kaldemar: show me a way to get the current SIP firmware for the Polycoms using Google and I'll be impressed ;) |
| 20:03.13 | [TK]D-Fender | rhombus: http://www.google.ca/search?hl=en&q=polycom+SIP+software+releases+matrix&btnG=Search&meta= |
| 20:03.19 | [TK]D-Fender | rhombus: FIRST LINK |
| 20:03.26 | [TK]D-Fender | winds up for the pitch |
| 20:03.29 | *** join/#asterisk AndreasDG (n=andreas@c85-196-92-50.static.sdsl.no) |
| 20:03.46 | AndreasDG | Hello! |
| 20:03.50 | kaldemar | ruben23: sorry, what? |
| 20:03.53 | rue_mohr | dont know how you found it, polycom dosn't list the 601 ontheir site |
| 20:04.07 | AndreasDG | Does anyone have any experience with asterisk and Cisco 7941 IP- Phones? |
| 20:04.33 | ruben23 | <PROTECTED> |
| 20:04.44 | rhombus | [TK]D-Fender: Awesome! Are these publicly available now? |
| 20:05.09 | rhombus | [TK]D-Fender: They used to provide them to "authorized partners only." |
| 20:05.20 | [TK]D-Fender | rhombus: Yes |
| 20:05.40 | [TK]D-Fender | rue_mohr: SEARCH BOX |
| 20:05.56 | rue_mohr | mumbles |
| 20:05.57 | rhombus | [TK]D-Fender: Amazing! What made them remove their head from their ass? |
| 20:05.59 | AndreasDG | noone? |
| 20:06.09 | kaldemar | rue_mohr: 601 is discontinued, but they do list it. |
| 20:06.31 | rue_mohr | just a sec |
| 20:06.34 | [TK]D-Fender | rue_mohr: http://search.polycom.com/query.html?charset=utf-8&la=en®ionlang=%2Fusa%2Fen&col=usaen&style=zusaen&qt=IP+601+user+guide&submit.x=0&submit.y=0&submit=search |
| 20:07.07 | rhombus | AndreasDG: Well, I don't :) |
| 20:07.17 | kaldemar | rhombus: check qualify and qualifyfreq in the sample sip.conf. |
| 20:07.37 | rue_mohr | http://www.polycom.com/products/voice/desktop_solutions/soundpoint/index.html |
| 20:07.41 | rue_mohr | no 601 |
| 20:07.44 | rhombus | kaldemar: wow. are you answering a message I sent last week? |
| 20:08.07 | [TK]D-Fender | rue_mohr: As you were told, DISCONTINUED. Go look for Model-T info on Ford's page wihle you're at it. |
| 20:08.08 | kaldemar | rhombus: sorry, wrong address. :) |
| 20:08.15 | rue_mohr | yea my point was thats were I was looking |
| 20:08.27 | kaldemar | ruben23: check qualify and qualifyfreq in the sample sip.conf. |
| 20:08.29 | rhombus | kaldemar: I did ask a question last week to which that would have been the answer, though :P |
| 20:08.30 | rue_mohr | that and google |
| 20:08.36 | [TK]D-Fender | rue_mohr: yes, your search and alpha-waves stopped there too... |
| 20:08.42 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 20:09.02 | WHYS | Is there a good *-video how-to somewhere? I want to demo video for my boss, but I am off to a slow start. (1.6 and probably x-lite) |
| 20:09.05 | [TK]D-Fender | unplugs the life-support machinery |
| 20:09.06 | kaldemar | rhombus: haha, now you'll remember those options years from now. :) |
| 20:09.33 | [TK]D-Fender | WHYS: Go look on the WIKI... its 3 options for SIP in sip.conf |
| 20:09.53 | WHYS | ok, Thanks |
| 20:09.58 | rhombus | Well, I'm thrilled that Polycom has found a clue. They finally got back to me about partner status 18 months after I applied. I guess they've figured out that playing fortress doesn't help sales. |
| 20:10.27 | *** join/#asterisk asheron (n=unamei@a194-109-2-226.dmn.xs4all.nl) |
| 20:11.09 | ruben23 | if my asterisk server is behind nat....should i set to add "nat=yes" on my sip usr and for the VOIP context...? |
| 20:11.37 | Qwell | Mattchis: Don't do that. |
| 20:11.40 | Khratos | Are there substancial changes on Manager interface between Asterisk 1.4.22.1 and Asterisk 1.4.23.1 ? I ask because a software I did on php worked just fine making Asterisk edit its own config files, but now I receive a lot of 'broken pipe' errors |
| 20:11.41 | [TK]D-Fender | ~sipnat |
| 20:11.41 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:11.46 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 20:11.47 | [TK]D-Fender | ruben23: READ ^^^^^^^^^ |
| 20:15.39 | Khratos | Are there substancial changes on Manager interface between Asterisk 1.4.22.1 and Asterisk 1.4.23.1 ? I ask because a software I did on php worked just fine making Asterisk edit its own config files, but now I receive a lot of 'broken pipe' errors |
| 20:18.59 | Khratos | utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 20:20.46 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:23.10 | *** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com) |
| 20:23.38 | [TK]D-Fender | Whee! Microsoft announces Windows 7 editions -- THERE'S 6! |
| 20:24.51 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 20:25.26 | beek_ | yawns |
| 20:28.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:30.38 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 20:31.59 | jjshoe | hrm when I do a sip debug on an itsp, I see in the contact section a local ip, not the boxes externalip, thoughts on how I can change that? |
| 20:33.40 | jjshoe | externip no? |
| 20:34.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:34.14 | Jeff_Phillips | hi |
| 20:34.25 | keith4 | [TK]D-Fender: wow... you've been busy |
| 20:34.27 | keith4 | makes popcorn |
| 20:38.27 | rhombus | [TK]D-Fender: Are there any good reasons to upgrade to the 3.1.1 Polycom SIP image from 2.1.1? Everything I have deployed is working. |
| 20:38.59 | jjshoe | right now I have it shown Contact: <sip:s@theexternalip> will the s affect anything? |
| 20:39.06 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 20:40.13 | Jeff_Phillips | i'm assuming that this failure to detect that the caller hung up occurs more often but that us hanging up the extension phone triggers it to disconnect the call anyway |
| 20:40.17 | Jeff_Phillips | right? |
| 20:40.35 | Jeff_Phillips | oops I thought I was typing in the other channel (sorry) |
| 20:40.46 | Jeff_Phillips | but actually it's just as relevant here |
| 20:41.05 | Jeff_Phillips | incomming analog zap channel hits voice mail, and fails to ever notice that the caller hung up so it just keeps repeating "if you'd like to review your message" |
| 20:42.34 | [TK]D-Fender | rhombus: You can read the changelogs as well as I can copy/paste them.... you tell ME |
| 20:43.13 | rhombus | [TK]D-Fender: Just seeing if you have an opinion, and changelogs can lie :) |
| 20:43.24 | [TK]D-Fender | rhombus: Not Polycom's :) |
| 20:43.29 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:43.30 | rhombus | [TK]D-Fender: But whatever -- sorry to put you out |
| 20:44.10 | [TK]D-Fender | rhombus: lot of cool stuff added... YMMV depending on model(s), etc |
| 20:47.15 | vader-- | tkd any experience with hooking security/fire alarm panels in with asterisk? |
| 20:47.34 | [TK]D-Fender | vader--: DON'T |
| 20:47.45 | vader-- | this ata has a fax pass through setting, was wondering if that helps or hinders |
| 20:47.56 | vader-- | ya i kinda need to though |
| 20:48.09 | [TK]D-Fender | vader--: No, you don't |
| 20:48.58 | vader-- | i have a remote location that all i have going to it is data/fiber line |
| 20:49.36 | [TK]D-Fender | vader--: Well i guess you'd better reconsider then. |
| 20:51.41 | *** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it) |
| 20:51.57 | jjshoe | hrm vitelity and this setup is driving me nutts. |
| 20:52.29 | keith4 | vader--: sounds like you need a cell setup |
| 20:52.48 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 20:56.24 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 20:56.42 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 20:58.43 | sipy | microsoft sqlserver MSDE vista |
| 20:58.53 | sipy | arrggghh!! |
| 21:01.10 | *** join/#asterisk Valmon (n=m_dorset@viliar.static.corbina.ru) |
| 21:01.23 | Valmon | Hello to All! |
| 21:01.31 | AndreasDG | o.O |
| 21:01.50 | Valmon | mmm? |
| 21:02.25 | Valmon | One,probably not simples at least for me question |
| 21:03.17 | Valmon | I want to separate sip provider in two conext. in and out |
| 21:03.29 | Valmon | *context |
| 21:03.48 | Valmon | *sorry for my english* |
| 21:04.11 | Valmon | I have used google. |
| 21:04.15 | [TK]D-Fender | Valmon: Provider's don't HAVE an "out" context. your PHONES do. |
| 21:04.22 | Valmon | But find only to topics. |
| 21:04.27 | Valmon | Yeah |
| 21:04.34 | Valmon | I understand it |
| 21:05.26 | Valmon | It's important for me: Calls from provider to me come in one context |
| 21:05.38 | *** join/#asterisk agx (n=badpengu@88-149-224-96.dynamic.ngi.it) |
| 21:05.40 | [TK]D-Fender | Valmon: You set that context in their peer entry |
| 21:06.00 | Valmon | Hm. |
| 21:06.01 | agx | Does 1.6 support PRESENCE SIP messages? |
| 21:06.16 | Valmon | Language barier ) |
| 21:06.22 | agx | ehm... sorry... i meant PUBLISH :) |
| 21:06.40 | [TK]D-Fender | agx: * sends presence, but does not receive it. |
| 21:06.54 | Valmon | Not a problem to put calls from provider to some context |
| 21:07.12 | Valmon | But it really problem, to put in the same time |
| 21:07.21 | Valmon | calls to him to another context. |
| 21:08.03 | Valmon | Yes, I can move in dialplan call from one context to another |
| 21:08.08 | Valmon | but. |
| 21:08.32 | agx | [TK]D-Fender, ok thanks |
| 21:08.35 | Valmon | Let's imagine |
| 21:08.50 | Valmon | I call from my sip-provider on my mobile phone |
| 21:09.02 | *** part/#asterisk agx (n=badpengu@88-149-224-96.dynamic.ngi.it) |
| 21:09.08 | *** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 21:09.11 | *** join/#asterisk riddlebox (n=user@75-105-81-181.cust.wildblue.net) |
| 21:09.12 | Valmon | After that I want to transfer call from mobile phone to some internal context |
| 21:09.39 | Valmon | but i get error no such extension |
| 21:09.48 | ariel_ | hello folks |
| 21:09.50 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 21:10.31 | Valmon | yes, because such extension isn't int context from sip.conf |
| 21:10.48 | Valmon | In sip.cong we have tip2 |
| 21:10.59 | Valmon | Tip 2: Use separate type=peer and type=user sections for SIP providers |
| 21:11.11 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 21:11.24 | Valmon | And this configuration almost unworkable at least for me |
| 21:11.36 | Valmon | At the googling I fount two topics |
| 21:11.55 | mcargile | is there a way to test the accuracy of res_timing_pthread.so in asterisk 1.6 like dahdi_test? |
| 21:12.02 | Valmon | http://osdir.com/ml/telephony.pbx.openpbx.users/2007-05/msg00034.html |
| 21:12.29 | Valmon | At the end of this topic -- wa only one advice. Use type=friend |
| 21:12.32 | `paul | can i have a database (my own ... (mysql)) containing numbers.. and asterisk reads from that database and does not allow calls (outbound ) to those numbers? |
| 21:12.40 | Valmon | so I can forget about separating |
| 21:12.58 | deadpigeon | Valmon: your seperate sip provider for your "out" context is simply an outbound route if im not mistaken |
| 21:13.24 | Valmon | Yes, it work, then I call |
| 21:13.40 | Valmon | But then I want to return by transfer call from called phone |
| 21:13.41 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 21:13.50 | mcargile | `paul: yes you can through dialplan the dialplan and an agi script |
| 21:13.52 | Valmon | I get error about extentions |
| 21:14.05 | Valmon | it search not in output context |
| 21:14.13 | Valmon | but in context from sip.conf |
| 21:14.44 | Valmon | That's the point |
| 21:14.45 | deadpigeon | im not seeing the advtange in having two seperate sip providers. |
| 21:15.02 | deadpigeon | of course i only use pri trunks anyways. |
| 21:15.04 | mcargile | `paul: you would have a catch all extension that would call the agi script. The agi script would do the look up in the database. If the number is not in the database it would do a goto to a context that actually can dial out |
| 21:15.05 | Valmon | Ok |
| 21:15.30 | mcargile | `paul: if it was in the database you could hang up or play a message |
| 21:15.31 | Valmon | Let's imagine -- you want to transfre call to external employer |
| 21:15.37 | Valmon | on mobile phone |
| 21:15.55 | LemensTS | http://pastebin.com/m33b6692d 41283 is correct, but i cant figure out how to save it as a variable...can someone help me? |
| 21:16.02 | Valmon | How he can return call to another internal number? |
| 21:16.49 | Valmon | internal extentions don't appear in incoming context, isn't so? |
| 21:17.11 | Valmon | It's alla about security |
| 21:17.17 | deadpigeon | hm. |
| 21:17.27 | Valmon | So |
| 21:17.41 | Valmon | If we can separate context _at_ the sip.conf |
| 21:17.49 | Valmon | it's configuration is real |
| 21:18.25 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 21:19.23 | deadpigeon | I wish I could help you more, but I guess I'm at a loss as to how to go about doing it. I've more experience with gr303's and the last 2 weeks have been a crash course in asterisk. |
| 21:19.34 | deadpigeon | Sorry. Hopefully someone else here has an idea? |
| 21:19.49 | Valmon | Yeah, I understand |
| 21:20.32 | [TK]D-Fender | LemensTS: Save it as a variable? Huh? |
| 21:20.45 | Valmon | I found working example, but is really disaster... |
| 21:20.46 | [TK]D-Fender | LemensTS: Right now you have basic variable use in PHP issues. |
| 21:21.04 | [TK]D-Fender | LemensTS: Nowhere are you setting and * var while would be with another AGI call. |
| 21:21.27 | [TK]D-Fender | an* |
| 21:21.35 | *** join/#asterisk nightrid3r (n=kvirc@78-20-228-200.access.telenet.be) |
| 21:22.28 | *** join/#asterisk zamba (i=marius@sveigde.hih.no) |
| 21:22.38 | zamba | is it possible to have more than one register line? |
| 21:23.02 | [TK]D-Fender | zamba: Sure |
| 21:23.31 | *** part/#asterisk tjfontaine (n=tjfontai@oftc/staff/tjfontaine) |
| 21:23.49 | Valmon | So separating provider to two context isn't so simple.... I really wonder, what it's rarelly used... |
| 21:23.58 | zamba | [TK]D-Fender: i'm trying to set up a sip trunk here.. it works perfectly for incoming calls, but i'm unable to dial out.. what in sip.conf is the problem? is it the [peer] declaration, the register line or both? |
| 21:24.21 | [TK]D-Fender | zamba: registering has NOTHING to do with placing calls |
| 21:24.25 | [TK]D-Fender | ~sipregister |
| 21:24.26 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 21:24.32 | kannan | hello , asterisk + iaxmodem with hylafax finally working fully :) |
| 21:24.42 | riddlebox | i hate waiting onhold at least its a company using asterisk |
| 21:24.48 | zamba | [TK]D-Fender: ok, so register is for incoming calls, period..? |
| 21:25.03 | [TK]D-Fender | zamba: Yes |
| 21:25.09 | zamba | [TK]D-Fender: but the context declaration under the peer declaration says something about which context to place incoming calls in, or am i mistaken? |
| 21:26.25 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 21:27.04 | *** join/#asterisk boynas (n=garyflor@wsip-98-190-136-194.ph.ph.cox.net) |
| 21:29.51 | zamba | [TK]D-Fender: no? |
| 21:32.16 | boynas | I have a question: theres this setup of over 100 phones that will possibly groww to 200. I am distributing the phones in 3 different servers. The switched network is only for voip traffic, all three servers connect to PSTN via IAX to a media gateway that has the pris. I am using polycom phones. Is there a max amount of polycoms suggested for one network? Is there a better way to do this?... Thanks |
| 21:36.46 | zamba | i'm able to dial outbound by using three pieces of information in ekiga, username, password and sip proxy/registrar.. but when i try to replicate this setup in my asterisk, i'm unable to get my calls through |
| 21:37.02 | zamba | can someone help me set up asterisk for a sip trunk this way? |
| 21:41.11 | LemensTS | TK: http://pastebin.com/m50880412 i verbosed the array and it shows 'data' is blank...i see it says result = 0, but before it said result = 1...its not looking at the current array? im not sure what you mean by "Nowhere are you setting and * var while would be with another AGI call" |
| 21:41.25 | *** join/#asterisk morganx (n=morgan@pool-71-191-101-193.washdc.east.verizon.net) |
| 21:42.38 | *** join/#asterisk saftsack (n=oliver@e179221147.adsl.alicedsl.de) |
| 21:44.15 | *** join/#asterisk jcape (n=jcape@209.120.251.66) |
| 21:45.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:46.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:50.52 | *** join/#asterisk docelm0 (n=chatzill@206-248-239-194.unassigned.ntelos.net) |
| 21:54.58 | docelm0 | Anyone know why a sip peer having insecure=very would be prompted to authentication in the sip messages? |
| 21:54.58 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 21:55.37 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 21:56.34 | DavidR2008 | I hope this question isn't off topic, if it just let me know, any able to offer any aastra 9133i config help? I can't get it to register with my asterisk server |
| 21:56.45 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 21:56.45 | DavidR2008 | *anyone |
| 21:57.19 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 22:00.21 | sipy | docelm0: to allow registered hosts to call without re-authenticating |
| 22:06.37 | *** join/#asterisk [netman] (n=netman@244.Red-79-145-182.dynamicIP.rima-tde.net) |
| 22:06.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:10.08 | jaytee | quittin time, be back later |
| 22:11.25 | boynas | I have a question: theres this setup of over 100 phones that will possibly groww to 200. I am distributing the phones in 3 different servers. The switched network is only for voip traffic, all three servers connect to PSTN via IAX to a media gateway that has the pris. I am using polycom phones. Is there a max amount of polycoms suggested for one network? Is there a better way to do this?... Thanks |
| 22:12.54 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 22:13.00 | [TK]D-Fender | boynas: For the scale you've mentioned I see no need for multiple servers or networks |
| 22:15.08 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 22:17.17 | *** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br) |
| 22:18.39 | *** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br) |
| 22:18.39 | DavidR2008 | I get 401 Unauthorized when my phone tries to register to my server. However I don't see anything else that helps me figure out what went wrong. I've double checked the phone's configuration and my sip.conf file. |
| 22:19.27 | *** join/#asterisk javb (n=javb@tdev213-76.codetel.net.do) |
| 22:19.39 | javb | is it posible to install Asterisk without installing zaptel ? |
| 22:19.56 | Qwell | javb: Do you have hardware that requires zaptel? |
| 22:20.15 | DavidR2008 | javb: sure, as long as you don't have hardware that requires zaptel |
| 22:20.20 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:21.58 | javb | I dont have the hardware. |
| 22:21.58 | [TK]D-Fender | DavidR2008: Your auth is wrong |
| 22:22.12 | javb | But i though that i needed the ztdummy to emulate the clock... |
| 22:22.58 | [TK]D-Fender | javb: You only need zaptel for hardware that uses it or MeetMe / IAX Trunking |
| 22:23.08 | javb | So, there is no timming if there is no zaptel. |
| 22:23.10 | DavidR2008 | I recognize that's what it means, but I've double checked (it's been many more times then two now) both the phone config and the sip.conf file and they have the same information. |
| 22:23.24 | javb | I need the ZAPTEL for anything that requires the timer, lime Meetme and IAX2 trunking? |
| 22:24.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:24.32 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.98) |
| 22:24.34 | [TK]D-Fender | javb: What I just said |
| 22:24.53 | edwin_quijada | What kiind of IP phone to use for a PBX |
| 22:25.11 | [TK]D-Fender | edwin_quijada: Whatever you want that speaks a protocol that * does |
| 22:25.12 | DavidR2008 | I'm able to register with a soft phone, so it's more of an aastra question I guess |
| 22:26.43 | *** join/#asterisk mrsci (n=sq@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
| 22:29.24 | edwin_quijada | [TK]D-Fender: Imean about brand or quality |
| 22:29.35 | [TK]D-Fender | edwin_quijada: Polycom > All |
| 22:29.50 | jjshoe | Qwell spent forever fucking with someone's itsp, rebooted router and poof it worked, then they where kind enough to state 3rd party firmware, bleh. |
| 22:34.01 | *** join/#asterisk bgmarete_ (n=marebri_@196.201.208.129) |
| 22:35.02 | *** join/#asterisk citywok (n=chatzill@corpnet.csgopenline.com) |
| 22:36.29 | *** join/#asterisk Get_The_Fish (n=IceChat7@75.151.94.189) |
| 22:39.58 | edwin_quijada | somebody has tested the IP Phone 7940G Cisco with Asterisk? |
| 22:44.07 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 22:46.11 | [TK]D-Fender | edwin_quijada: Cisco = pricey & trouble |
| 22:49.09 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:49.53 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:50.48 | *** join/#asterisk [netman] (n=netman@96.Red-83-45-36.dynamicIP.rima-tde.net) |
| 22:50.48 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 22:52.02 | *** part/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 23:02.01 | *** join/#asterisk RobertLaptop (n=rmiddle@68-244-121-71.pools.spcsdns.net) |
| 23:05.42 | [TK]D-Fender | BBIAB |
| 23:08.59 | rue_mohr | reads user guide for phone |
| 23:12.10 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 23:21.48 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 23:28.33 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 23:32.24 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-3bc283879a698854) |
| 23:37.09 | *** join/#asterisk emrahpbx (i=MRH@f210085.upc-f.chello.nl) |
| 23:39.26 | talirk81 | in an AGI (php for example) is there a clean way to pull all the $agi_XXXX varibles into an array? such as $AGI['callerid'] or $AGI['callderidname'] |
| 23:44.46 | *** join/#asterisk MrNeutr0n (n=DrLexus@209-253-217-62.ip.mcleodusa.net) |
| 23:45.24 | MrNeutr0n | greetings everyone - i was wondering if maybe any of you have had trouble of a similar sort as I have |
| 23:46.04 | MikeJ | wow.. quiet in here |
| 23:46.49 | rue_mohr | hmm is the telemarketer torture come with asterisk? |
| 23:47.00 | MikeJ | smacks file upside the head and runs |
| 23:47.02 | MrNeutr0n | well, in case anyone has any ideas, here is my question |
| 23:47.23 | MrNeutr0n | I keep getting "all circuits busy" on outbound calls over my zap trunks |
| 23:47.35 | Qwell | MikeJ: We're sorry, the file you have attempted to smack is not here right now. Please leave a message and file will get back with you as soon as possible. |
| 23:47.45 | MikeJ | :( |
| 23:47.50 | emrahpbx | MrNeutr0n: do you also receive an error like: 500 interna server error ? |
| 23:47.50 | MrNeutr0n | with this message: chan_zap.c: Not yet hungup... Calling hangup once with icause, and clea |
| 23:47.50 | MrNeutr0n | ring call |
| 23:47.51 | MikeJ | heya Qwell |
| 23:47.54 | MrNeutr0n | hahahahah |
| 23:47.55 | Qwell | MikeJ: hey |
| 23:48.09 | MrNeutr0n | yeah i understand - so where should i start? |
| 23:48.19 | Corydon76-dig | MrNeutr0n: PRI trunks? |
| 23:48.31 | MrNeutr0n | Corydon76-dig, yes |
| 23:48.53 | Corydon76-dig | MrNeutr0n: Check that your switchtype is correct |
| 23:49.08 | Corydon76-dig | MrNeutr0n: also, I'd suggest upgrading to a version with chan_dahdi |
| 23:49.09 | MrNeutr0n | and each "hangup with icause" is surrounded by an AUDIO MODE value: ON(1) |
| 23:49.12 | *** part/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 23:49.16 | MrNeutr0n | and AUDIO MODE vOFF(0) |
| 23:49.25 | MrNeutr0n | I was wondering about dahdi |
| 23:49.39 | MrNeutr0n | unfortunately I think we're tied to this trixbox distribution |
| 23:49.44 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 23:50.09 | Corydon76-dig | Ah, that sucks |
| 23:50.13 | MrNeutr0n | Corydon76-dig, so would the problem with the switchtype result in intermittent errors? |
| 23:50.28 | MrNeutr0n | because it doesn't happen all the time |
| 23:50.33 | MrNeutr0n | and it never gets logged in the CDR |
| 23:50.46 | Corydon76-dig | Well, CDRs aren't for that |
| 23:50.59 | ACK-NAK | how to I update dahdi-linux:? Obviously, make, make install, then what? What is the minimally invasive way to reload dahdi without rebooting? |
| 23:51.01 | Corydon76-dig | messages log is where it would go |
| 23:51.10 | ACK-NAK | ...or must I reboot? |
| 23:51.25 | Corydon76-dig | ACK-NAK: modprobe -r zaptel |
| 23:51.25 | Qwell | ACK-NAK: stop asterisk, then /etc/init.d/dahdi restart |
| 23:51.41 | Qwell | ACK-NAK: That'll reload all the kernel modules. |
| 23:52.11 | MrNeutr0n | Corydon76-dig, ok, that's what I suspected - CDR is one step higher up the chart, so to say |
| 23:52.13 | ACK-NAK | Qwell: Much obliged! Is that documented somewhere? I was looking around for the answer. |
| 23:52.26 | Corydon76-dig | MrNeutr0n: well, no, it's a different type of logging |
| 23:52.28 | MrNeutr0n | messages log - ok i have just been looking in full |
| 23:52.31 | Qwell | ACK-NAK: it's a fairly standard way of restarting a daemon |
| 23:52.31 | ACK-NAK | Qwell: didn't want to trouble you good folks unnecessarily |
| 23:53.03 | Corydon76-dig | MrNeutr0n: messages or full, pretty much the same, except for debugging going to full |
| 23:53.11 | emrahpbx | how can i find the error which causes: "500 internal server error" on outbound calls, without nat. tried 5 /6 sip trunks, they all fail and on other servers with the same settings it works great...but since yesterday it doesnt work on my own server.... any tips ? |
| 23:53.32 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:53.45 | Corydon76-dig | emrahpbx: check that your password is correct, especially capitalization |
| 23:54.07 | emrahpbx | Corydon76-dig: password of the voip trunk ? |
| 23:54.26 | Corydon76-dig | emrahpbx: Nothing changed on your system? |
| 23:54.41 | Corydon76-dig | You need to call the support line of your voip provider, then |
| 23:54.54 | emrahpbx | Corydon76-dig: i don't remember i changed anything...ive removed 1.4 installed 1.6 / 1.7 and now 1.4 and problem exists.. |
| 23:55.14 | MrNeutr0n | so where can i find out what causes DIALSTATUS to sometimes be CHANUNAVAIL |
| 23:55.18 | MrNeutr0n | and other times CONGESTION |
| 23:55.25 | sipy | That would classify as at least a minor change |
| 23:55.25 | MrNeutr0n | both seem to lead to the same result |
| 23:55.43 | MrNeutr0n | Corydon76-dig, thanks for your help by the way |
| 23:55.56 | Corydon76-dig | emrahpbx: have you carefully read UPGRADE.txt ? |
| 23:56.00 | emrahpbx | Corydon76-dig: the same settings works on another system.. so, the voip provider is responding correctly. when i enter the user details in a voip client, it works also... |
| 23:56.29 | drmessano | 1.7? |
| 23:56.58 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 23:56.58 | jaytee | I was just wondering about that. They're holding out on us, keeping the good stuff just for the newbs. |
| 23:57.14 | emrahpbx | Corydon76-dig: i will have a look on it again, but its so weird... and i also see, when i install the manager-gui "127.0.0.1 cannot authenticate", user details are correct, everything is correct... this thing is really annoying |
| 23:57.22 | drmessano | jaytee: Actually, I was thinking "Thats old".. I am on 1.7.2 |
| 23:57.40 | jaytee | drmessano, yeah and that bong needs a rest! |
| 23:58.20 | *** join/#asterisk Pryon (n=Pryon@animalcules.com) |
| 23:58.27 | Corydon76-dig | the manager-gui? |
| 23:58.37 | emrahpbx | asterisk-gui * |
| 23:58.50 | jaytee | asterisk has a gui? |
| 23:58.57 | jaytee | I had no idea! |
| 23:59.16 | Corydon76-dig | Check that you've installed the version that matches with 1.6, if that's your underlying version |
| 23:59.18 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f82920213be7fb29) |
| 23:59.30 | Corydon76-dig | jaytee: yeah, it's the one developed for use with the AA50 |
| 23:59.42 | Qwell | Corydon76-dig: pretty sure he's being sarcastic |
| 23:59.49 | emrahpbx | Corydon76-dig: theres just one working asterisk-gui (2.0) right? |
