| 00:00.00 | Adolph-testing | i will try now |
| 00:00.04 | Olobola | does anyone know why my incoming calls are several hours off? The time on my linux machine is correct. |
| 00:00.22 | JT | because you're several hours off UTC? |
| 00:00.31 | JT | i assume you're talking about CDRs or something |
| 00:00.47 | Olobola | yes, cdr |
| 00:01.09 | JT | you must be a few hours off UTC |
| 00:01.47 | Olobola | if I run date it gives me the correct time. What should I do? |
| 00:01.50 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808846.dsl.bell.ca) |
| 00:02.20 | JT | i just told you the answer |
| 00:02.23 | JT | pay attention |
| 00:02.28 | JT | CDRs are recorded in UTC |
| 00:02.32 | JT | this is normal |
| 00:02.36 | Adolph-testing | x-lite |
| 00:02.39 | Olobola | I see |
| 00:02.43 | Adolph-testing | i got same error 404 |
| 00:03.02 | JT | Adolph-testing: sounds like the provider sucks |
| 00:03.29 | Olobola | I see. So when I parse my logs I should just adjust the time -+ |
| 00:03.33 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 00:04.05 | Adolph-testing | fuck .. :( i just buyet 1 year contract for small bussinees |
| 00:04.05 | JT | you might be able to record custom CDRs if this is a problem |
| 00:04.15 | Adolph-testing | what they need to change on they server? |
| 00:04.36 | [TK]D-Fender | JT, sounds like all the settings are being changed at the same time so there's no hope of finding the right combination. thats what happens when the lock keeps changing behind your back. |
| 00:04.52 | JT | Adolph-testing: they need to accept your auth for one |
| 00:05.02 | JT | [TK]D-Fender: which settings? |
| 00:05.29 | JT | Adolph-testing: hrm, probably a good idea to test a voip provider before signing up to a big contract |
| 00:05.32 | [TK]D-Fender | Adolph-testing, Tell them to set a normal user & pass and give them to you. THEN setup a soft-phone and keep working on that that until IT works. THEN work on Asterisk. Not before. |
| 00:06.03 | [TK]D-Fender | JT, auth on ITSP side. he's messing his confis locally, they're changing stuff on the server and spinning around each other in circles. |
| 00:06.30 | JT | ag |
| 00:06.51 | Adolph-testing | i will do this [TK]D-Fender |
| 00:11.00 | Yourname`` | fujin : makes sense |
| 00:13.28 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 00:13.56 | km- | Hey, do I want the unsigned sip firmware or the signed sip firmware from cisco's site? I want to upgrade to 8.8 but not sure which version to get. |
| 00:14.07 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 00:14.09 | [TK]D-Fender | Yourname``, No, it doesn't. If they see the transferrer's CID then you should stop doing ATTENDED transfers, and start doing BLIND transfers |
| 00:15.17 | fujin | bxfer on my 942's still shows the transferrs cid |
| 00:15.56 | Yourname`` | [TK]D-Fender: I wish I could do that. :) |
| 00:16.25 | *** join/#asterisk exothermc (n=miles@izetta.office2-ww.wideideas.net) |
| 00:16.47 | [TK]D-Fender | Yourname``, and the reason you can't is.....? |
| 00:17.28 | Yourname`` | [TK]D-Fender: because these stupid phones are fuct, and there is a NEED for an attended transfer. Hand off of a lot of important information. |
| 00:17.42 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:18.02 | km- | Note: You must install Cisco Unified Call Manager version 5.0 or higher to utilize the 8.8 SIP firmware image. |
| 00:18.03 | km- | hmm. |
| 00:18.09 | [TK]D-Fender | Yourname``, TFB <- |
| 00:18.12 | km- | that obviously negates my question entirely. |
| 00:18.23 | [TK]D-Fender | Yourname``, Start attended, cancel, return blind. |
| 00:18.46 | [TK]D-Fender | km-, That makes no sense. |
| 00:19.18 | km- | from cisco's release notes, no less. |
| 00:19.26 | [TK]D-Fender | km-, unless they mean that only CCM5+ can actaully take full advantage of your phone+SIP8.8 |
| 00:19.46 | [TK]D-Fender | km-, that may be a warning on CCM's behalf more than the phone. |
| 00:19.52 | km- | eh, I don't know for sure. |
| 00:19.59 | [TK]D-Fender | km-, only way that adds up |
| 00:20.24 | km- | the firmware files are a different format as well |
| 00:20.34 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 00:20.37 | km- | cmterm-7940-7960-8.7.00-sip.cop as opposed to .sbn's |
| 00:21.02 | Yourname`` | [TK]D-Fender: Vertical 9133is make it hard to do all that :S |
| 00:21.25 | [TK]D-Fender | Yourname``, Learn how to use your own phones. |
| 00:22.04 | km- | you know, I have 8.2 on this phone. I just need to remember how to fix this problem where it endlessly RRQ's for the conf file |
| 00:22.14 | km- | I know I fixed it once before... |
| 00:22.20 | Yourname`` | [TK]D-Fender : ok |
| 00:25.25 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 00:26.00 | exothermc | Is there a good way to migrate from file based voicemail to IMAP? |
| 00:26.56 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 00:29.39 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 00:30.41 | Yourname`` | What's a good tftpd server for centos? |
| 00:31.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 00:33.30 | exothermc | yum install tftpd ? |
| 00:33.58 | Yourname`` | doesn't work. |
| 00:34.30 | exothermc | yum install tftp-server ? |
| 00:34.42 | Yourname`` | That's bringing home xinetd too. |
| 00:36.21 | fujin | most wil |
| 00:36.23 | fujin | will* |
| 00:36.26 | fujin | that's why I use atftpd |
| 00:36.37 | km- | heh, I'm installing atftpd now |
| 00:36.38 | fujin | you can launch it standalone, and from (x)inetd |
| 00:36.48 | Yourname`` | I hate xinted |
| 00:37.39 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 00:37.43 | Adolph-testing | JT: u can tell me how to configure asterisk server to accept incoming sip connections? |
| 00:38.01 | Adolph-testing | amin to make a sample config |
| 00:38.02 | Yourname`` | I think atftp-server would do it, fujin |
| 00:38.21 | fujin | That's probably a centosism |
| 00:39.00 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3a12ae42de64862b) |
| 00:39.34 | *** join/#asterisk RoyK (n=roy@91.149.31.174) |
| 00:42.48 | Yourname`` | Yeah, prolly. |
| 00:43.11 | *** part/#asterisk RoyK (n=roy@91.149.31.174) |
| 00:43.27 | JT | Adolph-testing: why are you asking me? |
| 00:44.22 | [TK]D-Fender | Adolph-testing, Do you now have a softphone fully working with your provider? |
| 00:51.21 | *** join/#asterisk Adolph-testin (n=andreiu_@2.128.219.87.dynamic.jazztel.es) |
| 00:52.33 | Adolph-testin | yes [TK]D-Fender i have xlite but i`m trying now to configure my asterisk server to try to connect with xlite on it and to see how it work and to give infos to the voiceral support to know how to make my sip account to work |
| 00:53.01 | km- | heh. it was my freaking nat setup that was screwing with my tftp. bugger. |
| 00:53.10 | Adolph-testin | Your technician made a very good point, download the xlite softphone (it's free) and input the information we have given you |
| 00:53.18 | [TK]D-Fender | adolph-testin : Stop wasting time and follow the instructions you've been given. |
| 00:54.00 | Adolph-testin | i followed but no working u think if that instruction are good i was here to ask you guys ? |
| 00:54.01 | [TK]D-Fender | adolph-testin : Setup X-Lite to communicate DIRECTY with your provider. FORGET about Asterisk until that is 100% SUCCESSFUL. |
| 00:55.06 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 00:55.07 | Adolph-testin | i set up man with user pass and voip proxy that voiceral support gived me but NO WORKING the problem is from him server and they don`t know to make it to woek |
| 00:55.08 | Adolph-testin | :( |
| 00:55.58 | [TK]D-Fender | adolph-testin : STOP TALKING ABOUT ASTERISK. Get it working with your softphone FIRST. You are wasting time! |
| 00:56.25 | [TK]D-Fender | adolph-testin : Go prove that your account is FUNCTIONAL with a simple tool you shouldn't screw up configuring. |
| 00:56.43 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 00:56.57 | Adolph-testin | yes man this is what i do, i am trying to register xlite on voiceral network but i get 404 error |
| 00:56.59 | Adolph-testin | :( |
| 00:57.32 | [TK]D-Fender | adolph-testin : then eitehr their account isn't setup right or you are entering it in wrong (for the 4 fields it takes). |
| 00:57.57 | [TK]D-Fender | adolph-testin : Go ask them to help you set up X-Lite. If you can't get that working then you are screwed. |
| 00:58.30 | Adolph-testin | u don`t understand me, they don`t know how to do this |
| 00:59.38 | [TK]D-Fender | adolph-testin : And you aren't listening... I said ASK YOUR PROVIDER. if they can't help you with this then you are beyond help. |
| 01:00.13 | Adolph-testin | MY PROVIDER DON~T KNOW TO SET UP MY SIP ACCOUNT |
| 01:00.49 | [TK]D-Fender | adolph-testin : You saying they can't walk you through setting up X-Lite? if so they are completely useless. Get a new provider. |
| 01:01.09 | Adolph-testin | but i paid this |
| 01:01.14 | [TK]D-Fender | adolph-testin : Thats like a mechanic who can't change spark-plugs. |
| 01:01.25 | Adolph-testin | this is cause i`m trying so more |
| 01:01.31 | Adolph-testin | yea |
| 01:01.37 | Adolph-testin | like this |
| 01:01.38 | Adolph-testin | :( |
| 01:01.50 | [TK]D-Fender | adolph-testin : Go get your money back. |
| 01:02.15 | Adolph-testin | u know a good voip provider that support unlimited channels ? |
| 01:02.54 | [TK]D-Fender | adolph-testin : Don't bet on "unlimited" what do you truly NEED? |
| 01:02.59 | fujin | lol yeah |
| 01:03.01 | fujin | why unlimited |
| 01:03.07 | fujin | you can't even work out how to use one ;) |
| 01:03.32 | [TK]D-Fender | ... |
| 01:04.05 | Adolph-testin | or with 100 minimum |
| 01:04.25 | [TK]D-Fender | adolph-testin : What kind of connection do you have? |
| 01:04.44 | Adolph-testin | i have a server with 10 mb broadband |
| 01:04.48 | [TK]D-Fender | adolph-testin : And why do you need so many channels? |
| 01:05.02 | Adolph-testin | i have a telemarketing system |
| 01:05.18 | fujin | lol |
| 01:05.21 | fujin | a telespamming system? |
| 01:05.21 | fujin | :) |
| 01:05.26 | fujin | man I wrote one of those the other dray |
| 01:05.28 | fujin | day |
| 01:05.35 | fujin | for some dude in here |
| 01:05.54 | [TK]D-Fender | adolph-testin : Here's what you need to do : |
| 01:05.57 | [TK]D-Fender | ~hafc |
| 01:05.58 | jbot | i guess hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 01:05.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 01:06.20 | craigk | hmmm - did the peaceful creaters of Asterisk intend it to be used for such evil as telespamming ? ;) |
| 01:06.31 | [TK]D-Fender | adolph-testin : Go check out the WIKI for a list of people to choose from. |
| 01:06.32 | [TK]D-Fender | ~wikis |
| 01:06.33 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 01:06.34 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 01:06.35 | fujin | yay~ he left |
| 01:06.37 | fujin | oh my god |
| 01:06.44 | fujin | it was that andreiu guy |
| 01:06.46 | fujin | who ripped me off |
| 01:07.13 | fujin | cunt! |
| 01:07.13 | [TK]D-Fender | fujin, wait, wasn't he the one who had "blackslk" as a partner in here last week or so? |
| 01:07.22 | fujin | I don't know. |
| 01:07.33 | fujin | I *WROTE* the telespamming stuff for him. |
| 01:07.34 | [TK]D-Fender | fujin, I think I know them... |
| 01:07.34 | JT | fujin: someone ripped you off in here? |
| 01:07.38 | fujin | Yes. That guy. |
| 01:07.45 | JT | fujin: he didn't pay? |
| 01:07.47 | fujin | He paypalled me $850USD from frauded accounts. |
| 01:07.52 | JT | wtf |
| 01:07.53 | Qwell_ | "that" guy? |
| 01:07.53 | fujin | which was promptly frozen. |
| 01:07.56 | fujin | *that* guy! |
| 01:08.01 | Qwell_ | oh, blackslk |
| 01:08.02 | fujin | 14:06:27) • Quits: Adolph-testin (n=andreiu_@2.128.219.87.dynamic.jazztel.es) : [ ] |
| 01:08.42 | fujin | little bastard |
| 01:08.57 | JT | 10:40 <Adolph-testing> hey please help me |
| 01:08.57 | JT | 10:41 <JT> please don't pm m |
| 01:08.57 | JT | 10:41 <JT> me |
| 01:08.57 | JT | 10:41 <Adolph-testing> sorry but i really need help |
| 01:08.58 | JT | 10:42 <JT> sorry but there's an irc channel, unless you're paying for consulting |
| 01:09.05 | JT | that was about 1.5 hours ago |
| 01:09.10 | fujin | Yeah |
| 01:09.14 | fujin | I got a similar message last week or whatever |
| 01:09.19 | fujin | told him a similar thing |
| 01:09.23 | fujin | which led onto me consulting for him |
| 01:09.25 | [TK]D-Fender | fujin, Ditto. |
| 01:09.31 | fujin | and being frauded out of a few hours work |
| 01:09.34 | [TK]D-Fender | (minus taking them on. |
| 01:09.52 | fujin | heh |
| 01:09.53 | JT | fujin: at least it should give you a little satisfaction that he's been unable to use it |
| 01:09.58 | fujin | Ha~ |
| 01:10.00 | fujin | Indeed. |
| 01:10.11 | fujin | The code wasn't exactly advanced |
| 01:10.14 | fujin | but shit |
| 01:11.16 | km- | I seriously need to find some decongestant. Stat. |
| 01:11.17 | fujin | half of it was from something I googled, then made it forking |
| 01:11.19 | fujin | http://homepages.maxnet.co.nz/~djfu/dial-threaded.pl.txt |
| 01:12.32 | km- | fujin: interesting script |
| 01:12.41 | fujin | heh :) |
| 01:12.59 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:13.04 | fujin | $850usd worth of frauded perl there |
| 01:13.11 | km- | oy. |
| 01:13.24 | km- | you should have known when he offered $850 for a two-page perl script |
| 01:13.43 | fujin | we initially agreed on 250 up front, 250 on completion |
| 01:13.48 | fujin | then the 250 on completion turned into 600 |
| 01:13.52 | fujin | from a different paypal account |
| 01:13.57 | JT | he must've found more accounts |
| 01:13.58 | fujin | both accoutns were apparently his "customers" |
| 01:14.15 | km- | nice, so he was using hacked paypal accounts to pay you? |
| 01:14.18 | km- | thereby you get nothing? |
| 01:14.53 | km- | Oh, is there anyone here who lives in the philadelphia area who wants a full time job managing a network of asterisk servers for a contact center |
| 01:15.08 | km- | the company tried to poach me but I told them I'd at least try to help them find someone else. :) |
| 01:16.00 | fujin | km-: aye, hacked accounts, paypal froze |
| 01:16.01 | fujin | so I get squat |
| 01:16.06 | km- | fujin: that blows. |
| 01:16.11 | fujin | unless they mysteriously believe him and unfreeze the transactions |
| 01:16.14 | fujin | yeah. |
| 01:16.17 | km- | I always provided services for asterisk in exchange for bartered hardware |
| 01:16.22 | km- | got a bunch of 7960's that way |
| 01:16.49 | km- | I think I traded 24 hours of professional services for a single 7960, worked out to both parties benefits |
| 01:18.56 | *** join/#asterisk ez` (n=ez@c75.152.78-116.clta.globetrotter.net) |
| 01:26.44 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 01:26.44 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 01:30.20 | *** join/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 01:39.03 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:39.42 | exothermc | Anyone know of a good way to migrate voicemail from file system to IMAP? |
| 01:40.11 | exothermc | or was IMAP designed only for people newly adopting asterisk and not for the existing user base? |
| 01:41.15 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:43.44 | *** part/#asterisk dijungal (n=kdaniel@205.244.149.157) |
| 01:49.03 | fujin | exothermc: I don't believe there are ny existing tools for mid-production migration |
| 01:52.15 | exothermc | hmm that was well thought out. |
| 01:53.30 | fujin | What was? |
| 01:53.37 | fujin | your mid-production migration? |
| 01:53.46 | exothermc | lack of. |
| 01:54.14 | fujin | I'm sure it'd be possible to write a migration tool, though. |
| 01:54.23 | fujin | start imap voicemail up, record a fiew voicemail messages |
| 01:54.29 | fujin | see what it does to each users maildir |
| 01:54.47 | fujin | then replicate that with your existing voicemail files, converting, mimencoding where necessary |
| 01:55.00 | exothermc | I would be very surprised if anything wasn't possible with enough effort. |
| 01:56.18 | fujin | so, exert the effort required? :) |
| 01:59.42 | [TK]D-Fender | exothermc, don't burn yourself out..... |
| 02:00.31 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 02:12.10 | Yourname`` | I can't believe this.. I've been trying to setup a tftp server on CentOS for about 45 mins now! And haven't been able to connect to it from win32 |
| 02:13.08 | *** join/#asterisk etfonhomey (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
| 02:14.39 | *** join/#asterisk Oztzrf (i=Oztzrf@adsl-76-214-7-77.dsl.lsan03.sbcglobal.net) |
| 02:21.48 | *** join/#asterisk _stink_ (n=stink@adsl-75-45-68-102.dsl.sfldmi.sbcglobal.net) |
| 02:23.05 | _stink_ | hi all - please let me know if I should ask this elsewhere: I have 6 SIP hardphones inside of a nat router, all using a trixbox server living at a CoLo. The phones keep losing touch with the server for incoming calls. Any advice on what settings I should use on the phone? e.g., use a random port, send keep-alive UDP packet every 20 seconds, use STUN, etc.? |
| 02:23.25 | _stink_ | and i've asked in #trixbox to no avail... and came here next |
| 02:26.38 | fujin | uhm |
| 02:26.54 | fujin | I'd personally run a sip gw inside the nat router |
| 02:26.56 | etfonhomey | Maybe qualify=yes in sip.conf for each of the phones entries? |
| 02:26.57 | fujin | and let that deal with keepalives etc |
| 02:27.53 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 02:32.01 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:41.24 | _stink_ | ok - thanks much for the suggestions, I'll try them out |
| 02:46.20 | JT | _stink_: qualify=yes, register regularly |
| 02:47.43 | Yourname`` | [OFFTOPIC] After I connect an Aastra 9133i to eth1, and then from eth2 I connect to the PC, how do I access it's webclient? |
| 02:49.16 | fujin | oh |
| 02:49.25 | [TK]D-Fender | Yourname``, Have you considered using a web-browser and looking at its IP? |
| 02:49.27 | fujin | forward packets between the interfaces? |
| 02:49.35 | Yourname`` | [TK]D-Fender: Yes I have |
| 02:49.39 | fujin | [TK]D-Fender: seperate networks one would hope |
| 02:49.47 | fujin | Yourname`: is there a gateway between the two networks? |
| 02:49.48 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 02:49.56 | _stink_ | JT: thanks - what's regular? like every minute? that doesn't interfere with current calls, right? |
| 02:50.05 | Yourname`` | fujin: I can't tell.. I'm asking someone over there to help me out. |
| 02:50.07 | fujin | _stink_: I have mine re-register every 30 seconds here |
| 02:50.19 | fujin | Yourname`: check the configuration on the nic dude |
| 02:50.40 | MrTelephone | yournme, whats your problem today? |
| 02:50.49 | _stink_ | fujin: ah, cool. |
| 02:50.57 | fujin | _stink_: with no adverse affects, if anything, positive affects (when one server fails, the heartbeat server kicks in 10~ seconds later, phones re-register another 15-20 seconds after the second server is up) |
| 02:51.07 | fujin | minimal downtime |
| 02:51.24 | Yourname`` | MrTelephone: A LOT Of problems, trying to setup a tftp server on centos, AND trying to get into Aastra 9133i's webinterface to download it's .cfg and then set'em up on the tftp server. |
| 02:52.48 | MrTelephone | tftp should be a snap? |
| 02:52.48 | _stink_ | fujin: nice. thanks much |
| 02:52.48 | Yourname`` | fujin: I mean, from the wall jack.. it goes to one eth port on the aastra, and the other eth port takes another eth cable to the computer. |
| 02:52.48 | Yourname`` | MrTelephone: I KNOW! |
| 02:52.52 | MrTelephone | im looking at my t1 pricing |
| 02:53.06 | fujin | Yourname`: eh, what? |
| 02:53.06 | Yourname`` | Ok, finally got the webclient iunterface now.. but its asking for a password, and I dont know the defaults |
| 02:53.09 | fujin | oic. |
| 02:53.11 | MrTelephone | a ds-1 is 450 each. if i buy 5 or more they are 240 each |
| 02:53.13 | fujin | admin/admin? :) |
| 02:53.20 | MrTelephone | ineed to start a call center or something |
| 02:53.20 | Yourname`` | nope |
| 02:53.25 | *** join/#asterisk mihinomenest (i=kgkJ@66.255.220.17) |
| 02:53.37 | fujin | how many channels is a ds1 |
| 02:53.40 | MrTelephone | how do you go about starting a call center? |
| 02:53.48 | fujin | that's 4 t1's, isn't it? |
| 02:53.50 | MrTelephone | 24 channels |
| 02:53.56 | MrTelephone | ds1 is a t1 |
| 02:53.58 | fujin | o_0 |
| 02:54.41 | JT | in PRI mode that would be 23 Bearer channels |
| 02:54.58 | fujin | pwnt by JT |
| 02:55.00 | fujin | ^5 JT! |
| 02:56.09 | Yourname`` | Ok, got the aastra part taken care of. |
| 02:56.20 | Yourname`` | Now, I just have to see if it'll connect to the tftp server |
| 02:57.35 | fujin | use your tftp connection abilities |
| 02:57.36 | fujin | they'll help |
| 02:57.57 | JT | fujin: well it could be in channelised mode |
| 02:59.43 | MrTelephone | what else can a guy do with 5 t1s |
| 02:59.46 | MrTelephone | :( |
| 03:01.05 | MrTelephone | how much is a dms100 |
| 03:01.08 | fujin | bond them together |
| 03:01.12 | fujin | browse the Intertron |
| 03:01.38 | Yourname`` | I downloaded local.cfg and server.cfg from the aastra |
| 03:01.47 | Yourname`` | server.cfg is empty, and local.cfg just has random stuff. |
| 03:01.50 | MrTelephone | i already have a 10mbit shaped oc |
| 03:02.03 | Yourname`` | I wonder why they dont have sample confs somewhere |
| 03:02.15 | MrTelephone | they probably do in the firmware upgrade zips |
| 03:03.48 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:03.53 | JT | MrTelephone: lots i bet |
| 03:04.22 | MrTelephone | u need money to make money :( |
| 03:05.16 | outtolunc | what is money? |
| 03:05.18 | JT | hundreds of thousands, i'm not sure |
| 03:05.52 | etfonhomey | Yourname`` Have you checked the wiki? |
| 03:06.33 | Yourname`` | etfonhomey: They have one? |
| 03:06.48 | etfonhomey | http://www.voip-info.org/wiki/index.php?page=Aastra+9133i+Configuration |
| 03:07.22 | Yourname`` | Oh, that one. Yessir. Did, but I'm looking to get one from the phone itself so I don't have to change a lot of stuff that way. |
| 03:08.02 | MrTelephone | http://www2.nortel.com/go/product_content.jsp?segId=0&parId=0&catId=-9224&prod_id=50103&locale=en-US |
| 03:10.31 | etfonhomey | That's the best I can do with Aastra's. But, if you had Polycom's... ;) |
| 03:11.02 | Yourname`` | hahaha |
| 03:11.05 | Yourname`` | I just might. |
| 03:17.33 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 03:19.40 | *** join/#asterisk hohum (n=dcorbe@wsip-70-166-81-42.sd.sd.cox.net) |
| 03:23.45 | Yourname`` | Ok, finally did it.. |
| 03:23.54 | Yourname`` | Now, I need to know if this phone is accessing the tftp server or not. |
| 03:26.37 | etfonhomey | Yourname'' Look at /var/log/messages |
| 03:27.12 | etfonhomey | Yourname`` Which tftpd are you using? |
| 03:27.29 | *** join/#asterisk Shihan (n=paulr@internet.sententia.com.au) |
| 03:27.55 | Yourname`` | tftp-server |
| 03:27.58 | Yourname`` | centos5 |
| 03:28.39 | Shihan | hi guys... small question... i want asterisk to install into /export/installed/asterisk-1.4.15, but it keeps trying to install things into /var/lib/asterisk... what configure switch controlls that? |
| 03:28.40 | etfonhomey | Let me check my config real quick. Is that the one that runs under xinet.d? |
| 03:29.35 | *** join/#asterisk mikecx (n=mikecx@cpe-76-181-117-188.columbus.res.rr.com) |
| 03:29.53 | [TK]D-Fender | Shihan, What OS? |
| 03:30.00 | Shihan | linux, fedora 8 |
| 03:30.12 | etfonhomey | Yourname`` Do you have a "tftp" text file under /etc/xinetd.d/ ? |
| 03:30.18 | [TK]D-Fender | Shihan, look at the Make options. |
| 03:30.32 | Yourname`` | yup |
| 03:30.38 | mikecx | if I used asteriskGUI to configure my outgoing lines (3 of them) will they have automatic rollover or do I need to setup trunking manually? |
| 03:30.52 | Yourname`` | Hold on, just noticed portsentry was blocking it. |
| 03:31.13 | Yourname`` | I stopped it, but I think the rule is still in iptables. |
| 03:31.15 | etfonhomey | Add this to the config: |
| 03:31.23 | etfonhomey | server_args = -vvvv -c -s /tftpboot |
| 03:31.34 | etfonhomey | But change tftpboot to the path to your tftp directory. |
| 03:31.43 | etfonhomey | Then you can look in the system log. |
| 03:32.13 | Yourname`` | done |
| 03:32.24 | Shihan | err.. by make options do you mean menuselect? |
| 03:33.37 | Yourname`` | Now I dont see anything in the logs at all :( |
| 03:33.42 | Yourname`` | I did an iptables -F |
| 03:33.53 | Yourname`` | And then tried, but nothing |
| 03:34.37 | etfonhomey | Restart xinetd? |
| 03:35.39 | etfonhomey | You got disable = no in your tftp file? |
| 03:36.32 | Yourname`` | Yup |
| 03:36.39 | Yourname`` | This is soooo messed up, lol |
| 03:38.02 | [TK]D-Fender | mikecx, This is not a GUI support channel, but * can roll-over you lines for OUTGOING. incoming is the responsibility of your telco to assign them in a "hunt group" |
| 03:38.16 | Yourname`` | I thought this would be the easiest thing ever.. |
| 03:38.40 | etfonhomey | I agree. |
| 03:38.50 | etfonhomey | You're not getting anything logged to /var/log/messages? |
| 03:39.19 | mikecx | [TK]D-Fender: Yeah, outgoing is what i'm worried about. I know the internal works properly (switching from BizPhone to asterisk). |
| 03:39.26 | fujin | 'stepping' is what some telcos call it, too |
| 03:39.34 | Yourname`` | Nothing at all. |
| 03:39.36 | fujin | Yourname`: why use xinetd to server tftp? |
| 03:39.40 | Yourname`` | I just setup tftp on another server.. |
| 03:39.47 | Yourname`` | fujin: BECAUSE CENTOS IS A POS! :( |
| 03:39.56 | Yourname`` | It runs it's tftp with xinetd |
| 03:40.03 | Yourname`` | That's my gripe too. |
| 03:40.04 | fujin | can you not run atftpd without xinetd? |
| 03:40.16 | Yourname`` | nope |
| 03:40.29 | Shihan | yeah ya can, its like tftpd -d or something |
| 03:40.48 | Yourname`` | etfonhomey: I even installed it to another server .. and now tried to connect to it there and it says cannot connect. It's ridiculous |
| 03:41.12 | [TK]D-Fender | mikecx, go learn about * and see how the configs are built. Many things in the GUI force to play by its rules and you make have toconfigure via them. |
| 03:41.21 | [TK]D-Fender | mikecx, For everything else theres : |
| 03:41.22 | [TK]D-Fender | ~book |
| 03:41.23 | 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 |
| 03:41.33 | [TK]D-Fender | mikecx, Get reading. |
| 03:41.44 | [TK]D-Fender | Yourname``, nothing wrong with CentOS. |
| 03:41.53 | Yourname`` | [TK]D-Fender: Nothing at all. |
| 03:41.54 | fujin | apart from the craptasticness |
| 03:41.55 | [TK]D-Fender | Yourname``, Works just fine |
| 03:42.09 | etfonhomey | Yourname`` Have you tried using a windows tftp client to send and/or retrieve a regular text file? |
| 03:42.22 | Shihan | in.tftpd -l is what you want if you dont want to run it in xinetd |
| 03:42.26 | Yourname`` | etfonhomey: Yes, and even that doesn't let me connect there. |
| 03:42.31 | [TK]D-Fender | It is by consequence the most well documeted distro out there and if you can't get it running a simple TFTP server well.... |
| 03:42.46 | mikecx | [TK]D-Fender: thanks, i've already kinda got an idea about most of the configs, it's just the fxo/fxs line rollover but i'll look through the book |
| 03:42.49 | Yourname`` | etfonhomey: This time, to make it easy on me I installed tftp on a very unrestricted box.. |
| 03:42.56 | Yourname`` | etfonhomey: And I dont see anything there either in the logs. |
| 03:43.06 | etfonhomey | Even things unrelated to tftp? |
| 03:43.46 | MrTelephone | could have installed debian 4.0 stable and typed apt-get install tftpd |
| 03:43.54 | etfonhomey | When it works you should see something like this: |
| 03:43.56 | MrTelephone | in the time it took you to get this to work |
| 03:43.59 | [TK]D-Fender | mikecx, If you want a preliminary opinion PASTEBIN your zapata.conf |
| 03:44.01 | [TK]D-Fender | ~pb |
| 03:44.02 | 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 |
| 03:44.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 03:44.13 | etfonhomey | messages.4:Nov 17 04:13:19 SYMLINUX in.tftpd[24869]: WRQ from 10.0.2.26 filename 0004f2130343-app.log |
| 03:44.24 | [TK]D-Fender | MrTelephone, oh please, this should have been a 5 minute job.... |
| 03:44.51 | MrTelephone | fenderbender |
| 03:45.07 | fujin | Dec 13 16:45:00 asterisk01 atftpd[26864]: Serving /spa942-000e08de22bb.cfg to 192.168.108.193:43453 |
| 03:45.11 | fujin | ;} |
| 03:45.27 | fujin | /usr/sbin/atftpd --daemon --port 69 --tftpd-timeout 300 --retry-timeout 5 --mcast-port 1758 --mcast-addr 239.239.239.0-255 --mcast-ttl 1 --maxthread 100 --verbose=5 /tftpboot |
| 03:45.30 | Yourname`` | etfonhomey: Nothing like it :( |
| 03:46.09 | MrTelephone | aftpd didn't work good for me through nat |
| 03:46.11 | etfonhomey | Yourname'' Pastebin your the tftp file from your /etc/xinetd.d directory. |
| 03:47.38 | Yourname`` | etfonhomey: http://pastebin.ca/814268 |
| 03:48.06 | mikecx | [TK]D-Fender: Guess i'm going to have to do that at work tomorrow, seems i'm pretty much firewalled out of my server there |
| 03:48.28 | MrTelephone | how did you firewall yourself out |
| 03:48.53 | etfonhomey | Yourname``, I don't have lines 15, 16, 17 in my file otherwise yours is identical to mine. |
| 03:49.08 | [TK]D-Fender | mikecx, productivity tip : Never ask for help when you can't follow through on it. |
| 03:49.42 | Yourname`` | yah |
| 03:49.54 | etfonhomey | Maybe pastebin your xinetd.conf as well. |
| 03:50.02 | mikecx | [TK]D-Fender: figured it would be a simple yes/no on whether or not the default setup would work. |
| 03:50.31 | [TK]D-Fender | mikecx, "default" and "GUI" and "asking in HERE?!" don't belong in the same sentence. |
| 03:51.14 | mikecx | [TK]D-Fender: asked in #asterisk-gui about 10 minutes ago (after you said this was the wrong place) but with only 30 total members, seems like no-one is there |
| 03:51.32 | [TK]D-Fender | mikecx, Feel like salmon yet? |
| 03:52.16 | fujin | SHIT CREEK |
| 03:52.37 | MrTelephone | no swearing |
| 03:52.37 | Yourname`` | etfonhomey: http://pastebin.ca/814275 |
| 03:52.52 | MrTelephone | how the hell do you firewall yourself out of your own server |
| 03:52.55 | mikecx | [TK]D-Fender: not really, though i was expecting the community to be a bit more... helpful though to be fair, not having access to the files sucks |
| 03:52.59 | MrTelephone | don't you have some back doors |
| 03:53.07 | mikecx | MrTelephone: it's a work server not meant to have internet access |
| 03:53.12 | MrTelephone | oh |
| 03:53.18 | fujin | use your VPN? |
| 03:53.18 | fujin | ;P |
| 03:53.21 | [TK]D-Fender | mikecx, Well... YOU can't help yourself any, don't think we're miracle workers. |
| 03:53.21 | mikecx | lol |
| 03:53.22 | MrTelephone | i need to read up on multicast |
| 03:53.33 | MrTelephone | do most routers block multicast? |
| 03:53.49 | MrTelephone | what stops someone from doing a massive multicast DOS |
| 03:53.54 | fujin | most cisco switches will allow moooolticast out of the box |
| 03:54.32 | etfonhomey | Yourname`` Here's mine: http://pastebin.ca/814280 |
| 03:54.42 | Shihan | why is the asterisk build doing this: http://pastebin.ca/814281 |
| 03:55.30 | Yourname`` | ok im officially giving up on this |
| 03:56.08 | etfonhomey | Did you try replacing yours with mine and then restarting xinetd? |
| 03:56.21 | fujin | MrTelephone: you can quite easily flood an upstream link if there is no moolticast router in the local subnet |
| 03:56.35 | fujin | as it will traverse default routes in order to find somethign which'll tell it how to route the multicast |
| 03:57.57 | Yourname`` | etfonhomey: lol no, lost the patience. :S |
| 03:58.03 | Yourname`` | Maybe when I'm not so pissed and tired |
| 03:58.20 | etfonhomey | Been there, done that... |
| 03:58.47 | MrTelephone | how come there isn't a way to multicast bittorrent :-P |
| 03:58.55 | Yourname`` | :( |
| 03:59.02 | Yourname`` | I'm really, really mad at this., |
| 03:59.04 | [TK]D-Fender | Shihan, go change the "makeopts" file to change you install paths and the call "make install" |
| 03:59.10 | fujin | It's not UDP? :D |
| 03:59.18 | etfonhomey | Yourname`` I think it's not even getting to your Linux box. |
| 04:00.05 | Yourname`` | probably.. |
| 04:00.18 | Shihan | ahhhh, much better, thanks for that... i thought that was build from configure? |
| 04:00.32 | MrTelephone | fujin, what kind of windows app will multicast? |
| 04:01.09 | fujin | norton ghost |
| 04:02.05 | MrTelephone | i wish that program could be setup for automatic imaging every morning or something |
| 04:02.13 | fujin | ha |
| 04:02.16 | fujin | that'd be handy. |
| 04:02.24 | MrTelephone | there are programs that do it but |
| 04:02.29 | MrTelephone | i don't know what they are called |
| 04:04.12 | [TK]D-Fender | G4L <--- |
| 04:04.35 | fujin | g4l :D |
| 04:04.39 | fujin | g4u actually bud |
| 04:04.46 | fujin | it's bsd, not leenux |
| 04:05.01 | Shihan | theres another one now too... called freeghost or something, its on sourceforge |
| 04:05.08 | [TK]D-Fender | http://sourceforge.net/projects/g4l <------ |
| 04:05.18 | [TK]D-Fender | fujin, O RLY? |
| 04:05.25 | fujin | ya indeed |
| 04:05.38 | fujin | g4u just STOLE THAT SHIT |
| 04:05.40 | fujin | nah, i dunno. |
| 04:05.45 | fujin | g4u is the original one ;P |
| 04:06.22 | Shihan | ahhh, fog is the one im thinking of |
| 04:06.35 | [TK]D-Fender | fujin, Yeah yeah... in in your day the greatest threat to man was swooping pteradactyls :p |
| 04:06.46 | fujin | piss off |
| 04:06.49 | fujin | i'm only 20 |
| 04:06.51 | [TK]D-Fender | pwned |
| 04:06.52 | fujin | ;> |
| 04:06.54 | Yourname`` | alright, good night guys. Apologize for my impatience and anger, I guess I'll try to tackle this tmrw with [TK]D-Fender's help |
| 04:07.04 | fujin | you're doing it wrong |
| 04:07.05 | fujin | get out -> |
| 04:07.13 | etfonhomey | l8r |
| 04:07.21 | [TK]D-Fender | ou812? |
| 04:07.30 | fujin | i10n |
| 04:07.44 | [TK]D-Fender | id10t |
| 04:07.51 | fujin | argh |
| 04:07.53 | fujin | my leg is so sore |
| 04:07.59 | [TK]D-Fender | TMI!@ |
| 04:08.03 | fujin | I pulled my groin muscle playing Hackey yesterday. |
| 04:08.06 | fujin | ;| |
| 04:08.53 | MrTelephone | gl4, it doesn't do live cloning? |
| 04:10.04 | Shihan | oky doky, thanks for the help guys... |
| 04:10.05 | [TK]D-Fender | fujin, future generations thank you! |
| 04:12.03 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 04:12.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:12.53 | *** join/#asterisk pepse (n=pepse@71-223-124-101.phnx.qwest.net) |
| 04:14.20 | MrTelephone | free ghost looks wicked |
| 04:21.41 | osiris | any idea why an inbound call would terminate after about 25 seconds ? |
| 04:22.13 | MrTelephone | the person hung up? |
| 04:22.20 | osiris | not one way audio, cause the inside extension dropped |
| 04:22.25 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 04:22.30 | tzanger | osiris: no audio stream, asterisk gives up after a while IIRC |
| 04:22.42 | MrTelephone | rtptimeout |
| 04:22.44 | osiris | not in this case |
| 04:22.53 | osiris | or so it seams |
| 04:23.02 | MrTelephone | what do the logs say |
| 04:23.14 | osiris | i can talk the whole time, and the polycom that is the extension just drops the call |
| 04:24.01 | osiris | nothing i can tell, but its a trixbox, and im kinda new to trixbox and asterisk. not new to linux |
| 04:24.27 | MrTelephone | you need to check your logs |
| 04:24.29 | MrTelephone | or sip debug |
| 04:24.43 | osiris | just got inbound.outbound working with NGT's broadsoft platform providing the trunk |
| 04:24.49 | mikecx | [TK]D-Fender: the book is useless for trunk groups |
| 04:25.05 | [TK]D-Fender | mikecx, Oh I don't buy that... |
| 04:25.17 | mikecx | The [trunkgroups] section is used for connections where multiple physical lines are |
| 04:25.17 | mikecx | used as a single logical connection to the telephone network, and won’t be discussed |
| 04:25.17 | mikecx | further in this book. If you require this type of functionality, see the |
| 04:25.18 | mikecx | zapata.conf.sample file and your favorite search engine for more information. |
| 04:25.19 | osiris | but the inbound calls all drop after 25 seconds. outbound works perfect |
| 04:25.25 | mikecx | sorry, shoulda used pastebin |
| 04:25.39 | [TK]D-Fender | mikecx, who said THAT is what you are looking for? |
| 04:25.46 | MrTelephone | osiris, is there 2 way audio> |
| 04:25.47 | MrTelephone | ? |
| 04:25.50 | osiris | could it be an inbound trunk setting causeing it to terminate ? |
| 04:25.54 | osiris | MrTelephone, oh yeah |
| 04:25.56 | [TK]D-Fender | mikecx, You just say a term you THOUGH meant something |
| 04:26.02 | [TK]D-Fender | (to YOU) |
| 04:26.07 | [TK]D-Fender | ~book |
| 04:26.07 | 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 |
| 04:26.09 | mikecx | [TK]D-Fender: sounds like it. multiple lines used as one |
| 04:26.13 | MrTelephone | if there is too much packet loss the call will drop |
| 04:26.38 | MrTelephone | mikecx.. group your zap channels then Dial(ZAP/g[group number]) |
| 04:26.41 | [TK]D-Fender | mikecx, that is for DIGITAL links to the telco. T1/E1/J1 |
| 04:27.09 | osiris | that shouldnt be the issue, and the extension is on the lan with the trixbox, and i have a pretty decent home network |
| 04:27.14 | mikecx | [TK]D-Fender: would be nice if they book would say that |
| 04:27.25 | osiris | good quality broadband |
| 04:27.34 | MrTelephone | mikecx, check your internet provider link.. cable modems might fail maintenance and it will be enough to knock out a call |
| 04:28.09 | MrTelephone | outbound works tho |
| 04:28.11 | MrTelephone | i forgot |
| 04:28.19 | osiris | totally |
| 04:28.40 | MrTelephone | mikecxx, tell qwell he needs to workon the documentation |
| 04:28.42 | MrTelephone | haha |
| 04:28.46 | osiris | call comes in, hits a ring group. answer the call, and it drops after 25 |
| 04:29.12 | MrTelephone | try asterisk instead of shitbox |
| 04:29.18 | mikecx | MrTelephone: lol, at this rate i'm just going to use guess and check, it'll be just as fast |
| 04:29.30 | osiris | most of my end users are going to be running trixbox |
| 04:29.34 | [TK]D-Fender | mikecx, So indeed you must set "group=1" (or a number from 0-31) and your dial should look like "Dial(Zap/g1/1234556645345)" substituting your group number |
| 04:30.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:30.23 | mikecx | [TK]D-Fender: thanks |
| 04:30.41 | MrTelephone | send fender 3.24 via paypal |
| 04:30.47 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 04:31.05 | MrTelephone | osiris, debug your sip messages |
| 04:31.29 | osiris | ill just run a sip trace tomorrow at the office |
| 04:31.36 | MrTelephone | osiris, your not going to get too far without some kind of error message |
| 04:31.43 | osiris | just wondering if something poped into anyones head |
| 04:32.14 | MrTelephone | that reminds me I have to do a trace on some dns srv problem I'm having |
| 04:33.01 | MrTelephone | who has a fast internet connection wand wants the movie superbad? |
| 04:33.22 | osiris | how fast, how big |
| 04:33.25 | osiris | what format |
| 04:33.34 | MrTelephone | some kind of avi 700 meg |
| 04:33.39 | MrTelephone | dvdrip |
| 04:33.43 | osiris | good quality ? |
| 04:33.46 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:33.56 | MrTelephone | I havn't watched it yet |
| 04:34.02 | MrTelephone | im just in one of those moods for sharing |
| 04:34.12 | osiris | i can get a meg a second off my torrents, so i'll take it. i ditched my cam rip |
| 04:34.17 | MrTelephone | its a dvdrip so it shold be good |
| 04:34.35 | MrTelephone | a MEG? |
| 04:34.37 | MrTelephone | yeah right |
| 04:34.41 | osiris | i got nuthin to do but kill BW and brain cells |
| 04:34.49 | MrTelephone | how do you get a meg |
| 04:34.50 | osiris | off my private site i do |
| 04:35.07 | osiris | tv and movies at 1.1 meg a second |
| 04:35.16 | MrTelephone | megabit or megabyte |
| 04:35.28 | osiris | comcast will burst 3 for about 20 seconds |
| 04:35.47 | MrTelephone | just one sec i'll be right back |
| 04:35.48 | osiris | byte i believe |
| 04:36.20 | osiris | idk, im going by what gkrellm is telling me. |
| 04:37.06 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 04:38.06 | osiris | its gonna time out |
| 04:38.11 | MrTelephone | i stopped downloading actual dvdrips because the divx rips are just as good |
| 04:38.31 | MrTelephone | mirc and their crappy dcc methods |
| 04:38.36 | MrTelephone | what a joke |
| 04:38.49 | MrTelephone | why do they time out |
| 04:38.57 | osiris | girewalls usually |
| 04:39.15 | MrTelephone | when is the last time someone sent you a dcc? |
| 04:39.30 | osiris | about 29 minutes |
| 04:39.38 | MrTelephone | for real? |
| 04:39.41 | osiris | yep |
| 04:40.01 | osiris | set your port range to something like 33310-33318 |
| 04:40.14 | osiris | some obscure high end range |
| 04:40.29 | osiris | make sure you forward yer ports too |
| 04:40.30 | MrTelephone | what is dcc passive i wonder |
| 04:40.38 | MrTelephone | i will if this doesn't work |
| 04:40.47 | osiris | much better |
| 04:41.17 | MrTelephone | oh shit i capped myself i forgot to ~t1 |
| 04:41.52 | osiris | need to restart it ? |
| 04:42.04 | MrTelephone | no we got one of those allot netenforcers |
| 04:42.11 | MrTelephone | just have to login and change a setting |
| 04:42.24 | gerphimum | hi everyone.. at work our phone system has a feature called reverse transfer. someone puts a call on hold at, say ext 1260.. then someone from ext 1240 (or some other phone) presses 4-1260 and is able to pick up the call that was placed on hold at the other phone.. im trying to recreate this behavior in asterisk but am not sure how. any suggestions ? |
| 04:42.44 | MrTelephone | its called parking |
| 04:42.52 | osiris | yep |
| 04:42.55 | gerphimum | ok. |
| 04:42.57 | gerphimum | ill look it up then |
| 04:42.58 | gerphimum | thanks |
| 04:43.07 | MrTelephone | i don't mean to sound rude |
| 04:43.21 | dexpdx | *yawn* |
| 04:43.23 | MrTelephone | but if you search park+asterisk you'll find it |
| 04:43.25 | gerphimum | hey, its a point in the proper direction ;) |
| 04:43.30 | MrTelephone | k |
| 04:46.14 | MrTelephone | can't get into my netenforcer for christ sakes |
| 04:48.35 | MrTelephone | i spent a couple hours today downloading a fireplace dvd for the tv |
| 04:48.42 | MrTelephone | could only find PAL versions |
| 04:49.13 | MrTelephone | converted it to NTSC and put 60 minutes of christmas tunes on it |
| 04:51.12 | osiris | nice work. i worked, and came home and started working again. |
| 04:51.34 | MrTelephone | typical IT lifestyle |
| 04:52.13 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:52.36 | osiris | yep |
| 04:52.52 | dexpdx | I hate how you can never easly look up the price of teleco equip |
| 04:52.59 | osiris | tryin to find wrt54G i can put dd-wrt on |
| 04:53.04 | MrTelephone | dex, same |
| 04:53.05 | dexpdx | you always gotta talk to some bozo for a "quote" |
| 04:53.22 | osiris | voipsupply.*** |
| 04:53.24 | dexpdx | So he can jack the price by 20Gs for commission |
| 04:53.25 | MrTelephone | dd-wrt? |
| 04:53.39 | osiris | linux firmware for linksys routers |
| 04:53.41 | MrTelephone | the model WRT54GL is what your looking for |
| 04:53.45 | osiris | nope |
| 04:54.02 | dexpdx | osiris: voipsupply doesn't seem to sell Avaya G860's |
| 04:54.12 | MrTelephone | the newet routers they started making the L series for people who want to change the firmware |
| 04:54.14 | osiris | the guy has a G, but its a version 6. which is a problematic hardware version |
| 04:54.17 | dexpdx | I just want to know how much the damn chassis cost |
| 04:54.34 | dexpdx | 54GL = magic |
| 04:54.36 | osiris | er i meant GL MrTelephone |
| 04:54.37 | dexpdx | works really well |
| 04:54.49 | MrTelephone | the older G ones work if they are old versions |
| 04:54.58 | dexpdx | dd-wrt + QoS for sip works pretty well for small to mid size office |
| 04:55.13 | MrTelephone | how much traffic can it handle? |
| 04:55.19 | osiris | its the only router i ever seen that when he makes a call from his ATA, the damn router (linksys wrt54GL) reboots |
| 04:55.20 | MrTelephone | im just gonna change my ip address |
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| 04:55.24 | dexpdx | MrTelephone: the dd-wrt? |
| 04:55.30 | MrTelephone | osiris, we'll restart the transfer in a sec |
| 04:55.37 | osiris | k |
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| 04:59.02 | MrTelephone | whats your connection at home osiris? |
| 04:59.27 | osiris | cable. |
| 04:59.42 | osiris | one sec. let me do a "right now" test |
| 05:00.05 | MrTelephone | its like your capped at 1meg or something |
| 05:02.18 | osiris | pm'ing results |
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| 05:30.37 | whymse | has anyone gotten a 79x1 cisco phone to register? I've gotten past the unprovisioned state, but I'm hung with JVM: %REG send failure: REGISTER |
| 05:41.20 | d-k-t | whymse, you checked out voip-info.org? |
| 05:41.47 | whymse | yeah |
| 05:42.14 | whymse | I'm using their config files.... everything should check out, but I'm not seeing the phone (7941G) hitting the SIP server via tcpdump |
| 05:43.07 | whymse | I was hoping someone may have gotten the permanent "registering" message like I am... there was one question with the same problem from Jun on the site, but sadly no answer |
| 05:43.38 | whymse | I have a 20 7940s working, so I'm not totally clueless |
| 05:44.02 | d-k-t | you using the SIP server IP in the 'callmanager' section? |
| 05:44.47 | whymse | in the processNodeName I have tried a hostname and IP of the SIP server |
| 05:45.02 | d-k-t | I've only used 7940s and 7960s myself, so I'm not much use there, short of being able to cast a second pair of eyes on what you've done to see if I can see anything obvious that doing it yourself you might miss :) |
| 05:45.42 | whymse | man, stick with the 79x0 and avoid the 79x1... they aren't worth the trouble |
| 05:46.36 | d-k-t | for the name are you using the fqdn? |
| 05:47.21 | whymse | yeah I have tried both fqdn and not... the comments on voip-info.org seemed to indicate I would end up unprovisioned if the callManagerGroup part was missing |
| 05:48.10 | whymse | it pulls the configs from the tftp server (which is the same box as the SIP server) |
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| 05:51.25 | d-k-t | whymse, you want to drop the config into a pastebin, minus any passcodes and fudging any names that you see fit? |
| 05:51.41 | whymse | yeah I might take you up on that... |
| 05:52.04 | whymse | Let me clean it up... I'm checking to see if the firewall may be clobbering UDP SIP requests on high ports |
| 05:52.58 | whymse | thanks for the help... btw |
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| 06:39.47 | gerphimum | can someone help me come up with an reason i can use to convince myself i need to buy some voip phones to play with ? |
| 06:39.57 | gerphimum | with a reason* |
| 06:44.28 | mikecx | gerphimum: because you have extra money and time and have nothing else better to do? |
| 06:46.30 | gerphimum | well, the last 2 for sure |
| 06:46.34 | gerphimum | extra money not so much |
| 06:46.55 | gerphimum | but ive been curious about this whole thing for quite some time now |
| 06:47.28 | mikecx | gerphimum: if you have a reason to want phone configuration, it's quite fun |
| 06:47.41 | gerphimum | well i dont really have an application for it, is all |
| 06:47.53 | gerphimum | i just wanna play with the system |
| 06:48.29 | mikecx | gerphimum: it is fun, but perhaps it might be better to spend the money elsewhere |
| 06:48.47 | gerphimum | probably so. |
| 06:48.59 | gerphimum | i could call it an investment.. |
| 06:49.00 | dexpdx | anyone else have problems with WaitExten inside a macro? |
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| 08:34.19 | awk | any ideas please! |
| 08:34.22 | awk | major break up |
| 08:34.23 | awk | <PROTECTED> |
| 08:34.36 | awk | any idea why my load is so high, but top doesn't show much processes using cpu, etc.. |
| 08:34.40 | awk | aswell as over a gig free of ram? |
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| 08:55.13 | tzafrir | awk, anything on top |
| 08:55.14 | tzafrir | ? |
| 08:55.27 | tzafrir | if not: look for processes in state 'D' |
| 08:55.50 | awk | tzafrir nothing in top |
| 08:56.00 | awk | and I cant use iotop and I dont have dtrace.. used this on solaris before |
| 08:56.12 | tzafrir | ps aux | grep D |
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| 08:58.30 | awk | http://www.pastebin.ca/814435 |
| 08:58.33 | awk | I cant see anything |
| 08:58.42 | awk | ignore the shutdown as I just issued that now and did a ps at that time.. |
| 09:00.06 | tzafrir | awk, there are several processes in D state. This is not a problem if it is for a short while |
| 09:00.17 | tzafrir | But if this stays for long - a big problem |
| 09:00.58 | tzafrir | I see 4 such processes in the the output |
| 09:01.23 | tzafrir | kjournald - a kernel thread. syslogd - the syslog daemon |
| 09:01.59 | tzafrir | Anyway, if this is the case, try asking in #$DISTRO |
| 09:02.11 | awk | k, thanks! |
| 09:02.15 | awk | you think its a centos bug? |
| 09:02.20 | awk | kernel bug? |
| 09:02.29 | tzafrir | filesystem problem? |
| 09:02.40 | awk | caused by? |
| 09:02.46 | awk | to larger log files, etc? |
| 09:04.36 | tzafrir | kernel or hardware problem, most likely |
| 09:04.43 | tzafrir | (if this is indeed the case) |
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| 09:07.55 | slavon_net | hello |
| 09:08.27 | slavon_net | how anyone tell to me how i can test that EXTEN exists in context? |
| 09:08.42 | slavon_net | ChanIsAvail(LOCAL/000${EXTEN}@office); allways retrurn 0 |
| 09:08.44 | awk | tzafrir any other ideas what could cause it |
| 09:08.46 | awk | or ways to determine it |
| 09:09.24 | tzafrir | not really |
| 09:09.34 | tzafrir | But you see those processes still in state D? |
| 09:09.46 | tzafrir | If syslog is hung, you now have no logs |
| 09:10.13 | juuva | anyone using vGSM? |
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| 10:09.23 | alphanet | hello. It looks like (1.2) Manager Interface cannot be used to start more than one originate: all the events are suspended til it ends. Is this the normal behaviour? I will for now revert to the "one manager interface connection per thread". |
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| 10:11.43 | FlatFoot | morning all |
| 10:12.34 | alphanet | oh, maybe Async can help me ... |
| 10:12.48 | FlatFoot | still got a prob can anyone help ? i need to declare a var to use within mutliple contexts BUT it needs to be a different value per context |
| 10:13.29 | FlatFoot | someone did say that i should try context_MYVAR=XXX , but that does not seem to work. Any ideas ? |
| 10:14.01 | FlatFoot | btw running version 1.4.11 |
| 10:20.13 | cjk | hi, i have one server with a digium card and i would that it forwards incoming calls to another server but in a way that this server has all the data so that rxfax could work. IAX and SIP are not appropriate for this. any other solution? some channel which forwards "raw" data |
| 10:21.42 | alphanet | cjk: IAX will forward raw data as long as you use the same codec (e.g. with ISDN: A-law or u-law depending on which side of the atlantic you are) |
| 10:22.37 | cjk | alphanet, and rxfax will work? what about sip? will it work with sip? |
| 10:22.53 | alphanet | cjk: should work with sip too, it's mostly a question of delay |
| 10:23.09 | alphanet | cjk: I have done it with servers on the same Ethernet and it worked (not worse than usual rxfax) |
| 10:23.24 | alphanet | cjk: however you might want to look into iaxmodem and hylafax |
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| 10:23.32 | cjk | alphanet, well, same switch. |
| 10:23.37 | alphanet | cjk: should work (tm) |
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| 10:23.44 | alphanet | cjk: as long as there is no codec conversion |
| 10:23.58 | b|uem00n | hello all |
| 10:24.24 | alphanet | cjk: or echo-cancel, etc |
| 10:24.43 | b|uem00n | anyone knows a method or a tutorial on testing the delay, jitter and packet loss on a network using open source or free tools? |
| 10:24.54 | cjk | alphanet, hmm, according to mailinglists wikis etc.... it should be luck when it works and should not be reliable.... |
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| 10:39.59 | alphanet | cjk: in general, faxing with tx/rxfax *is* unreliable |
| 10:40.05 | alphanet | cjk: apparently, iaxmodem is better |
| 10:40.18 | alphanet | cjk: now, if you add layers of IP networking, it will make things worse, yes. |
| 10:40.52 | alphanet | cjk: people usually recommend to do the fax -> file conversion at the borders of your network, then transmit using either files or T.110 (?) |
| 10:41.13 | cjk | alphanet, yep, but i would like to convert to a file on the second server not the first |
| 10:41.20 | cjk | gives me less troubles with db access etc.. |
| 10:41.27 | cjk | my pri server should just forward traffic |
| 10:41.36 | cjk | nothing else |
| 10:41.38 | cjk | no agi, no db |
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| 10:42.20 | alphanet | cjk: it should work, yes. |
| 10:42.36 | cjk | thanks for your opinion. i will give it a try |
| 10:43.26 | alphanet | :) |
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| 11:23.14 | Oerd | hi |
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| 11:23.44 | Oerd | anybody managed to get pickup working with mISDN and Snom telephones? |
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| 12:24.43 | MindTheGap | morning all... im trying to localize voicemail but its playing the wrong files. i've set language=pt_BR in sip.conf and voicemail.conf. Debug says SIP/600-08470c58> Playing 'vm-password' (language 'pt-BR') but it is actually playing /var/lib/sounds/vm-password not /var/lib/sound/pt_BR/vm-password |
| 12:30.03 | mosty | have you tried checking the debug log? |
| 12:30.21 | MindTheGap | i've tried to replace the files being played, but it seems asterisk wont play the right files cause it thinks it should be playing english format not pt_BR format, so, dates, enumeration, quantities and so on are all messed up... in pt_BR it should play INBOX.wav for one message and INBOXs.wav for more than one message, but it just plays INBOX.wav. |
| 12:33.18 | MindTheGap | Debug says SIP/600-08470c58> Playing 'vm-password' (language 'pt-BR') |
| 12:33.59 | MindTheGap | but its not playing /var/lib/sounds/pt_BR/vm-password |
| 12:34.18 | MindTheGap | is playing /var/lib/sounds/vm-password |
| 12:36.18 | mosty | and debug is set to 10 or something? |
| 12:39.29 | MindTheGap | same thing except it now says things like locked in /path/to/mailbox/, but nothing regarding language or path to files being played on voicemail |
| 12:39.49 | mosty | what version of asterisk are you running? |
| 12:39.55 | MindTheGap | 1.4.13 |
| 12:40.57 | mosty | i'd recommend trying 1.4.15, and if that looks the same, submit a bug report (after looking to see if there already is one) |
| 12:42.15 | MindTheGap | there's nothing on 1.4.13-1.4.15 changelog regarding changes in voicemail.conf except for imap thingies if i recall... |
| 12:42.55 | mosty | look for bug reports, then submit one if none exists |
| 12:44.30 | MindTheGap | any other things i should look for? language=pt_BR in other files, /var/lib/asterisk/sounds structure, anything? |
| 12:47.31 | mosty | i've only ever used english sound files, sorry |
| 12:48.21 | mosty | you could try strace'ing asterisk, and see if it tries to open files somewhere else but just not reporting it in the debug log |
| 12:48.43 | MindTheGap | hmm... and how do i do that? |
| 12:49.08 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 12:49.10 | stimpie | how can I get calls lasting 3 days when my dialplan includes set(TIMEOUT(absolute)=43200)? |
| 12:50.02 | mosty | MindTheGap, strace asterisk |
| 12:50.11 | mosty | redirect the output to a file |
| 12:50.19 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 12:57.16 | *** join/#asterisk BadBru (n=bad_b@ACA40FFD.ipt.aol.com) |
| 12:58.07 | BadBru | someone know how to install chanspy or zapbarge |
| 12:58.08 | BadBru | ? |
| 12:58.58 | BadBru | someone here ? |
| 13:00.11 | mosty | asterisk 1.4 has ChanSpy, nothing special is required beyond the standard install procedure |
| 13:00.33 | kaldemar | they both come with asterisk unless you scecifically don't install them. |
| 13:03.35 | BadBru | is there any relase for debian i could get with |
| 13:03.39 | BadBru | apt-get install ... |
| 13:05.26 | mosty | what debian release? |
| 13:05.44 | BadBru | actually... colinux |
| 13:05.59 | mosty | never heard of it |
| 13:06.11 | BadBru | it's somthing u can use on windows.. |
| 13:06.21 | BadBru | a.. "virtual debian linux" |
| 13:07.05 | mosty | no idea sorry, you're on your own with that |
| 13:07.07 | BadBru | i must pus in sources.list a link with asterisk |
| 13:07.20 | BadBru | like in any debian |
| 13:07.29 | BadBru | deb source file with asterisk lask version |
| 13:07.55 | mosty | debian unstable has asterisk 1.4, i have no idea if that will work with your dist |
| 13:08.34 | tzafrir | BadBru, colinux?? |
| 13:08.39 | BadBru | kernel 2.4.6-co0.6.1 |
| 13:08.40 | BadBru | yes |
| 13:09.30 | mosty | that is quite old. i would expect to have trouble with it |
| 13:09.49 | BadBru | what would u recomand me ? |
| 13:10.07 | BadBru | i want to use on win.. a wirtual linux.. |
| 13:10.57 | mosty | how about a regular linux dist in xen? |
| 13:11.47 | BadBru | i don't want another computer |
| 13:12.16 | *** join/#asterisk maache (n=maache76@82.101.148.50) |
| 13:12.22 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:12.29 | BadBru | wich is the older asterisk version wich have zapbarge or chanspy included by default ? |
| 13:13.18 | mosty | i doubt you can use zapbarge in a virtual machine |
| 13:13.41 | mosty | why don't you just compile asterisk 1.4 |
| 13:14.03 | tzafrir | BadBru, hmm... I don't know if anybody regularly builds it |
| 13:14.23 | tzafrir | BadBru, What version of Debian do you use? |
| 13:15.09 | tzafrir | And I suspect zaptel will not work nicely with it |
| 13:15.09 | BadBru | debian 4.0 |
| 13:15.19 | tzafrir | That's Etch (current Stable) |
| 13:15.25 | tzafrir | It has asterisk 1.2.13 |
| 13:15.29 | mosty | BadBru, i thought you said you were running colinux, not debian |
| 13:15.40 | tzafrir | Good enough to test basic things (such as: if zaptel works at all) |
| 13:15.50 | BadBru | colinux.. has mounted virtual debian 4.0 |
| 13:16.37 | tzafrir | For starters, apt-get install asterisk zaptel-source; m-a a-i zaptel |
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| 13:17.41 | BadBru | tzafrir let me get it.. zaptel is something like asterisk, or is an asterisk module ? |
| 13:18.06 | tzafrir | A hardware interface Asterisk uses |
| 13:18.06 | lirakis | BadBru: zaptel is a module for asterisk |
| 13:18.14 | tzafrir | And that is not part of the mainline kernel |
| 13:18.30 | lirakis | tzafrir: right.. sorry it is actually a kernel module |
| 13:21.11 | BadBru | on wich way can be usefull zaptel ? |
| 13:22.00 | lirakis | BadBru: for using TDM hardware, or for a dummy timing module in asterisk (for conferences etc.) |
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| 13:31.29 | mosty | woohoo, i finally solved my pri redirected call issue |
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| 13:33.04 | *** join/#asterisk dzalewski (n=dzalewsk@196.218.200.202) |
| 13:34.10 | dzalewski | Did anyone before integrated Panasonic KXTDA30 with Asterisk ? |
| 13:34.28 | [TK]D-Fender | dzalewski: Highly unlikely |
| 13:34.38 | dzalewski | I have 8 incoming line to the legacy PBX |
| 13:34.45 | dzalewski | I want to do an IVR on asterisk |
| 13:35.18 | dzalewski | I'm a bit confused about how many FXO/FXS ports I need :) |
| 13:35.31 | [TK]D-Fender | dzalewski: How would you have calls come in and out of * for this purpose? |
| 13:35.49 | [TK]D-Fender | dzalewski: And you do NOT want to use analog line ports for this. |
| 13:35.51 | dzalewski | I need to provide an IVR for all those 8 lines |
| 13:36.21 | dzalewski | PSTN -> Legacy PBX -> Asterisk(IVR) |
| 13:36.29 | dzalewski | how should I connect them ? |
| 13:37.21 | krdian_ | dzalewski: does it PBX have any voip card inside ? |
| 13:37.39 | dzalewski | krdian_: no it comes without any voip module |
| 13:38.20 | [TK]D-Fender | dzalewski: And how much would one cost? |
| 13:39.22 | *** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 13:39.31 | krdian_ | dzalewski: so you need to put PXO cards inside your * box |
| 13:39.34 | dzalewski | [TK]D-Fender: I guess it will be more cost effective to buy a FXO/FXS card :) |
| 13:40.03 | dzalewski | krdian_: yeah but I do not understand clear do I need FXO only or FXO and FXS |
| 13:40.16 | dzalewski | no analog or IP phones will be connected to the asterisk |
| 13:40.44 | mosty | dzalewski, you need FXO ports to connect to phone lines, and FXS ports to connect to the old PBX |
| 13:40.49 | mosty | (most likely) |
| 13:40.54 | krdian_ | dzalewski: so you can connect your box throug PX) cards to PBX |
| 13:41.10 | krdian_ | *FXO |
| 13:41.35 | dzalewski | mosty: All I need from * is to do an IVR |
| 13:41.42 | dzalewski | and leave panasonic like it is |
| 13:42.25 | [TK]D-Fender | dzalewski: and how do you expect * to send the call BACK into your system? |
| 13:42.39 | mosty | when i setup a sip phone to redirect to another number, how does that redirected call pass through my asterisk server's dialplan? |
| 13:43.28 | [TK]D-Fender | dzalewski: Are you planning on putting * in FRONT of your existing PBX, or BEHIND it? |
| 13:43.35 | dzalewski | [TK]D-Fender: behind |
| 13:44.23 | mosty | dzalewski, oh if you want asterisk behind the pbx, you would probably need an FXO port for asterisk. but that would only give you one line |
| 13:44.30 | [TK]D-Fender | mosty: forwarding is usually a SIP 302 redirect just takes the original channel and sends it to the target exten in your phone's context |
| 13:44.31 | krdian_ | mosty: as normal extension in phone context |
| 13:44.45 | [TK]D-Fender | dzalewski: So call comes in on your PBX, you push it out to * immediately? |
| 13:44.56 | mosty | my redirected calls appear to be using chan_local somehow |
| 13:45.07 | dzalewski | [TK]D-Fender: exactly, and then IVR will prompt |
| 13:45.13 | krdian_ | mosty: exactly |
| 13:45.14 | [TK]D-Fender | mosty: pastebin something useful. |
| 13:45.34 | [TK]D-Fender | dzalewski: Ok, so you have 8 ports to take your lines in from the telco, and 8 to go to *? |
| 13:45.58 | mosty | krdian_, but chan_local calls don't give me correct billsec in my CDR's, unless i add the /n option. how do i add that for these redirected calls? |
| 13:46.08 | steliosk | has anyone any experience (bad or good) with the OpenVox d110p card ? |
| 13:46.35 | dzalewski | [TK]D-Fender: correct |
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| 13:47.03 | [TK]D-Fender | dzalewski: Ok, what kind of ports are those on your Panasonic? are they CO ports, or analog station ports? |
| 13:47.27 | dzalewski | [TK]D-Fender: 12 analog ports |
| 13:47.35 | [TK]D-Fender | dzalewski: what KIND? |
| 13:47.48 | [TK]D-Fender | dzalewski: I'm talking about the 8 you would send to * |
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| 13:48.53 | [TK]D-Fender | dzalewski: You have 8 CO (to the telco) ports for your incoming line ports. What kind of ports are the ones going to Asterisk? Does it treat * like the telco, or like a phone? |
| 13:49.31 | dzalewski | [TK]D-Fender: like a phone |
| 13:49.43 | [TK]D-Fender | steliosk: Virtually no-one here would have anything to do with a clone card. It may work perfectly for you . YMMV. |
| 13:50.39 | [TK]D-Fender | dzalewski: Ok, then you can test this easily WITHOUT * at all. plug in a normal phone into that port. Have your system send the call out the port. Pick up the ringing phone and talk (pretending to be the IVR). how would that phone send the call somewhere else to free up the port? |
| 13:51.26 | krdian_ | mosty: hmmmm, strange i have to check this on my box |
| 13:52.36 | dzalewski | [TK]D-Fender: you mean that I will need 2 ports for each incoming line. one from panasonic pbx and one to panasonic? |
| 13:54.03 | [TK]D-Fender | dzalewski: No, I don't know the answer to this yet, although that is entirely possible. How would YOU do it? is there something you could do while holding that phone to transfer the call back into your system to to a person's desk phone for instance? |
| 13:54.54 | mosty | krdian_, [TK]D-Fender: http://pastebin.ca/814615 note that I need it to redirect to LOCAL/XXXXXX@outgoing/n |
| 13:55.33 | mosty | is it possible to set /n as default for all chan_local calls? |
| 13:57.06 | [TK]D-Fender | mosty: "/n" treats it as a 100% unique channel and doesn't clear upon reinvite. not sure if thats a good thing. 1 way I can think of is the set a "transfercontext" for the phone and then use a catch-all to nest another Local channel adding the "/n" |
| 13:59.05 | mosty | i use local channels elsewhere in my dialplan, and i have /n everywhere otherwise my cdr records don't give me what i want |
| 14:00.18 | mosty | how do i set the transfer context? i can't see any sip.conf settings |
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| 14:02.32 | mosty | ahh, i see it in channelvariables.txt - thanks |
| 14:03.16 | dzalewski | [TK]D-Fender: thanks, looks like I need to gather more info about this panasonic pbx cause I didn't see it in real. All I know is from it's manual :) |
| 14:09.32 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 14:10.57 | [TK]D-Fender | mosty: wiki up sip.conf |
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| 14:11.42 | jhb | [TK]D-Fender, just a quick feedback on yesterday - I did not get the originate for multiple calls working, but now pass a variable containing a list of numbers to it, and the local extension dials them in parallel. Decided thats more closely matching our use case |
| 14:11.59 | jhb | [TK]D-Fender, thanks a lot the help, it makes a huge difference |
| 14:12.18 | [TK]D-Fender | jhb: And the "cancel" feature... how'd you handle that? |
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| 14:12.41 | jhb | [TK]D-Fender, well, in the end it gets solved by dial - it drops the other ones. |
| 14:12.56 | jhb | [TK]D-Fender, I was postponing the information on that |
| 14:13.43 | jhb | [TK]D-Fender, so far, one way I found is basically using Set(DB(batch-groupid/orignumber)=${CHANNEL)) to have a lookup table |
| 14:13.55 | jhb | with groupid something known to all calls |
| 14:14.14 | jhb | that way I could do it with n originates as well |
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| 14:15.10 | jhb | [TK]D-Fender, its for the next version of directionless.info btw, will be documented and opensourced, so the help is not lost ;-) |
| 14:17.05 | [TK]D-Fender | jhb: Good to know. |
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| 14:17.41 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:22.50 | mocker | Anyone have any suggestions for queue monitoring software? |
| 14:22.59 | mocker | So a manager can see what's going on in the call queue. |
| 14:27.40 | mosty | mocker, i recommend you use asterisk realtime, and write your own |
| 14:27.56 | mosty | and i would implement my own queue's in agi or macro's |
| 14:28.16 | mocker | Eh, it's a pretty simple queue. |
| 14:29.24 | mosty | in my experience app_queue is so annoying, i have resolved never to use it again *shruh* |
| 14:29.36 | mocker | heh. |
| 14:29.44 | mocker | app_queue is... unique. |
| 14:30.46 | [TK]D-Fender | mocker: "monitor" in what way? |
| 14:31.44 | stimpie | I had a segmentation fault in asterisk, see full bt at: http://pastebin.com/m75bbb152 |
| 14:31.50 | mocker | [TK]D-Fender: Just who's on the phone and things like that. |
| 14:31.54 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 14:32.00 | mocker | Not getting into listening to calls yet. |
| 14:32.18 | [TK]D-Fender | mocker: There are tools ont he WIKI for that already, or write your own parsing out data from AMI |
| 14:32.32 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 14:32.52 | mocker | [TK]D-Fender: Right, but I was looking for apps recommended. |
| 14:32.54 | darkskiez | I remember seeing a script snippet that would allow you to call someone but allow them to press a key to accept the call or not. |
| 14:33.02 | mocker | Instead of installing 5 to see which one works well. :) |
| 14:34.00 | [TK]D-Fender | darkskiez: Thats jsut the "M" dial option for a "privacy" Macro. The catch with his is to do this across multiple simultaneous calls and let the others continue to have a chance to answer until an accept happens |
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| 14:34.04 | mosty | [TK]D-Fender, i'm using __FORWARD_CONTEXT, and calls get redirected to a new context i just created, is there a channel variable that i can use to find out the forwarding channel? |
| 14:34.37 | darkskiez | [TK]D-Fender: thats what i'm looking for .. yes |
| 14:34.37 | stimpie | any recommandations on my seg fault are welcome |
| 14:37.29 | [TK]D-Fender | mosty: no idea. |
| 14:38.26 | *** join/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 14:38.41 | yassine | hi everyone anyone here using asterik with ubuntu ? |
| 14:40.45 | Kobaz | oh noes |
| 14:40.53 | Kobaz | voicepulse completely flat outted |
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| 14:44.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:46.29 | stimpie | couldn't bug 10347 (crash using cdr_csv) happen in cdr_custom also? |
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| 14:59.51 | [TK]D-Fender | Kobaz: as in? |
| 15:00.02 | mosty | how can i handle billing cdr's that are the result of a forwarded call? a calls b and b forwards to c, asterisk 1.4.15 gives me three cdr's for this, but only a->b has billsec greater than 0 (I need the b->c billsec, which is currently 0) |
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| 15:07.36 | *** join/#asterisk twitchnln (n=twitch@cpe-orncorp.dktc.atl.oneringnetworks.net) |
| 15:07.55 | twitchnln | good morning, i have a * box that is frontending an altigen phone system via pri, my users are experiencing outbound dtmf problems (ie. sequence/lost digits in ivr) but when i am watching from the console, i see the digits coming in correctly, anyone got any idea where i should look to fix this? |
| 15:08.09 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 15:08.56 | mosty | are your users on sip phones? |
| 15:09.14 | mosty | try dtmf=auto in sip.conf |
| 15:09.16 | twitchnln | no, they are on altigen phones |
| 15:09.25 | e` | is there a way to reset someones voicemail recording to the default asterisk unavailable message? |
| 15:10.05 | mosty | twitchnln, pri for incoming and outgoing? |
| 15:10.13 | twitchnln | mosty: i have a pri on the back side and iax to pstn |
| 15:10.13 | [TK]D-Fender | twitchnln: "relaxdtmf=yes" |
| 15:10.48 | [TK]D-Fender | e`: Yes, delete the recording |
| 15:10.50 | twitchnln | [TK]D-Fender: tried that, no dtmf came across |
| 15:10.57 | [TK]D-Fender | twitchnln: Ew. |
| 15:11.21 | [TK]D-Fender | twitchnln: Next guess play with the gains a bit. |
| 15:11.35 | twitchnln | in zapata.conf? |
| 15:11.51 | twitchnln | or on the iax side? |
| 15:13.22 | [TK]D-Fender | twitchnln: PRI. |
| 15:13.48 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:14.06 | twitchnln | [TK]: will give that a shot, thanks |
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| 15:31.15 | Kobaz | [TK]D-Fender: oh, yeah umm, as in voicepulse is completely dead and unreachable |
| 15:31.34 | Kobaz | [TK]D-Fender: for about 20 minutes |
| 15:31.41 | *** part/#asterisk bkw__ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 15:31.42 | Kobaz | it's back up now, yay |
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| 15:58.11 | mosty | [TK]D-Fender, i figured out a workaround, CALLERID(rdnis) is set to the redirecting number, from there i can extract all the required info |
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| 16:02.37 | mocker | Is it possible to increase the amount of time Asterisk waits for digits during an attended xfer (using atxfer from features.conf)? |
| 16:04.32 | mosty | mocker, there are a few timeout options in features.conf - see if one of them does what you want |
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| 16:33.27 | darkskiez | I've called someone with a Dial M() macro, but if the callee hangs up the macro keeps going, where does it return to? |
| 16:33.29 | outtolunc | ~that is not true, if the world were round.. all ip phones would have IAX~ <G> |
| 16:33.29 | jbot | I think you lost me on that one, outtolunc |
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| 16:35.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:35.51 | outtolunc | darkskiez: original context iirc |
| 16:36.16 | outtolunc | http://lists.digium.com/pipermail/asterisk-users/2006-May/151755.html |
| 16:36.29 | darkskiez | outtolunc: thats for the caller :| |
| 16:37.05 | outtolunc | i seem to remember setting a macro and/or transfer context helps |
| 16:39.28 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 16:49.14 | *** join/#asterisk egypcio (n=egypcio@200.150.142.210) |
| 16:49.27 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:56.04 | *** join/#asterisk BCS-Satori (n=BCS-Sato@68.34.28.26) |
| 16:57.29 | *** join/#asterisk hohum (n=dcorbe@wsip-70-166-81-42.sd.sd.cox.net) |
| 16:58.39 | BCS-Satori | Morning, is there a way to send an alert to a receptionist, when a person off site launches their software sip phone which registers to asterisk. I need some way of alerting the receptionist, maybe email, or pbx some how alerting by a phone call, or something |
| 16:59.22 | BCS-Satori | almost like an alert for registering and unregistering of the client |
| 17:04.35 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:04.50 | blitzrage | BCS-Satori: well... you could use regexten and regcontext to place a NoOp() priority into asterisk, then have a script that parsed that context and basically did a diff between the last check, and the new check. Seems hackish... but is the first thing that came to mind |
| 17:04.56 | BadBru | some1 can help me with this error: |
| 17:04.56 | BadBru | -- Executing ZapBarge("SIP/300-0819b2a8", "1") in new stack |
| 17:04.57 | BadBru | Dec 13 12:01:17 WARNING[2578]: app_zapbarge.c:135 conf_run: Unable to open pseudo channel: No such file or directory |
| 17:05.14 | mosty | BCS-Satori, BLF/line presences? |
| 17:05.15 | blitzrage | BadBru: is zaptel/ztdummy loaded? (or some other hardware?) |
| 17:05.18 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 17:05.31 | BadBru | zaptel no |
| 17:05.32 | mosty | BadBru, are you running this on a virtual machine? |
| 17:05.35 | caio1982 | is anyone aware of any Asterisk event in new york city on january/february 2008? any user group meeting, maybe? |
| 17:05.37 | BadBru | yes |
| 17:05.46 | BadBru | not virtual |
| 17:05.52 | BadBru | on debian 4.0 |
| 17:06.38 | mosty | BadBru, do you have a TDM card? ZapBarge only works on Zap channels, and you only have Zap channels if you have a hardware interface to the phone system |
| 17:06.42 | BCS-Satori | mosty: we do use BLF on our SPA-932 on our phones but unfortantly they only show if the BLF is registered (green), perosn on phone (red), person's phone ringing (blinking red), there is no idication for when a phone isnt registered |
| 17:06.52 | BCS-Satori | blitzrage: ya, i see what you mean hehe |
| 17:07.03 | BadBru | no.. just want to listen sip calls |
| 17:07.48 | BadBru | no other hardware installed.. only iax.. and sip |
| 17:08.31 | tzafrir | BadBru, well, colinux is not exactly native |
| 17:08.46 | BadBru | hm.. |
| 17:08.57 | BadBru | it's not but it's working well |
| 17:09.30 | mosty | BadBru, well you can't use ZapBarge to listen to a SIP-SIP call |
| 17:09.43 | BadBru | then must use chanspy ? |
| 17:09.48 | BadBru | instead of zapbarge |
| 17:09.48 | mosty | yes |
| 17:10.38 | blitzrage | Zap == Zaptel == hardware |
| 17:11.04 | MindTheGap | hello ppl... sayunixtime(,,kM) is repeating the hour and adding a "minus" ... it says "minus fourteen, fourteen and five" |
| 17:11.16 | BadBru | fuck.. it's beeping ... |
| 17:12.02 | tzafrir | BadBru, what version of asterisk do you use? |
| 17:12.24 | BadBru | exten => 779,1,ChanSpy(scan/1000,80) |
| 17:12.36 | BadBru | 1.2.13 |
| 17:12.45 | BadBru | asterisk 1.2.13... |
| 17:13.05 | BadBru | what if i know the sip user i want to spyon |
| 17:13.21 | BadBru | how i know channel where sip user talking ? |
| 17:13.59 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:14.02 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 17:17.56 | blitzrage | BadBru: don't think the ability to listen to a specific channel was added until a later version (like 1.4+) |
| 17:18.00 | russellb | BadBru: you could start by using a version that isn't ancient |
| 17:18.04 | russellb | :) |
| 17:18.10 | blitzrage | I could be wrong.... but I haven't used 1.2 for more than a year now :) |
| 17:18.32 | [TK]D-Fender | BadBru: and how about reading.. the INSTRUCTIONS. |
| 17:18.35 | BadBru | ok then i will get it |
| 17:18.57 | BadBru | but it works.. chanspy.. but i don't dear anything.. only beep beep |
| 17:19.02 | russellb | 481 changes in asterisk 1.2 since 1.2.13 :) |
| 17:19.07 | russellb | <3 his changes_since script |
| 17:19.12 | blitzrage | [TK]D-Fender: you're like Austin Powers when he was having trouble controlling the VOLUME OF HIS VOICE |
| 17:19.25 | blitzrage | :) |
| 17:19.39 | [TK]D-Fender | blitzrage: No, that was entirely deliberate :) |
| 17:19.46 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.124) |
| 17:20.02 | blitzrage | I know... a bit too often in my estimation |
| 17:20.03 | tzafrir | Hey, give him a break. He way following the INSTRUCTIONS he got on a specific IRC channel (#asterisk) |
| 17:21.09 | BadBru | tzafrir, what is your problem ? u think ur smart ? |
| 17:21.11 | tzafrir | I actually find it strange that things work on colinux |
| 17:21.24 | BadBru | hm... |
| 17:21.28 | russellb | interesting bit of information ... we have made 1170 changes to asterisk 1.2 since 1.2.0, and 1723 changes to asterisk 1.4 since 1.4.0 |
| 17:21.52 | BadBru | if you have a logic expl not to work on colinux ? |
| 17:21.59 | Qwell | russellb: that's telling |
| 17:22.10 | BadBru | colinux load debian 4.0 |
| 17:22.21 | russellb | Qwell: i don't know what that means ... just found it interesting :) |
| 17:22.22 | BadBru | but when i give.. apt-get install asterisk |
| 17:22.33 | BadBru | it does not bring me asterisk 1.4 |
| 17:22.36 | Qwell | russellb: means we've been pwning :D |
| 17:22.42 | russellb | Qwell: w00t |
| 17:22.43 | BadBru | i will get it from download area.. |
| 17:22.59 | tzafrir | is colinux binary-compatible to i386 linux? |
| 17:23.53 | BadBru | yes |
| 17:24.07 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 17:24.08 | FlatFoot | blitzrage: think i might have found a bug with cdr_adaptive_odbc |
| 17:24.15 | hardwire | tzafrir: its quite neat |
| 17:24.24 | Yourname`` | Hello hello, errbody. Today, I feel well. |
| 17:24.33 | Corydon76-vcch | FlatFoot: oh? |
| 17:24.39 | FlatFoot | i'm using Freetds to connect to MSSQL and i have had to specify sending calldate |
| 17:25.20 | Corydon76-vcch | Interesting, that's one of the database I specifically tested |
| 17:25.20 | blitzrage | Corydon76-vcch: I'm gonna be testing func_odbc and ODBC VM storage with MS SQL for a client starting today... |
| 17:25.20 | tzafrir | if so, you can try deb http://pkg-voip.buildserver.net/debian etch main |
| 17:25.24 | FlatFoot | calldate didnot want to go via INSERT without being explicitly sent ie Set(CDR(calldate)= etc |
| 17:25.50 | BadBru | let's see |
| 17:25.50 | Corydon76-vcch | FlatFoot: did you set an alias in cdr_adaptive_odbc.conf ? |
| 17:25.59 | *** join/#asterisk ta^3 (n=tacvbo@1005hostc7.starwoodbroadband.com) |
| 17:26.11 | Corydon76-vcch | FlatFoot: calldate is NOT the name of the CDR internal field |
| 17:26.22 | FlatFoot | Corydon76-vcch: yes tried that but it did not work either |
| 17:26.26 | *** join/#asterisk ManxPower (n=manxpowe@246.sub-70-221-77.myvzw.com) |
| 17:26.53 | FlatFoot | really thats what i have in the standard table build script |
| 17:27.11 | Qwell | Corydon76-vcch: time for cdr_adaptive_psychic_odbc |
| 17:27.22 | Corydon76-vcch | FlatFoot: "alias start => calldate" is what you have? |
| 17:28.04 | Corydon76-vcch | Because the name of the field is "start" |
| 17:28.07 | FlatFoot | Corydon76-vcch: why start ? i thought calldate was standard . when i last used it with mySQL it was the case |
| 17:28.26 | Corydon76-vcch | FlatFoot: it is not the standard name, no |
| 17:28.38 | FlatFoot | ah sorry i was mistaken |
| 17:29.18 | MindTheGap | is there any way to call System() and get the output back to a variable? |
| 17:29.29 | FlatFoot | just that all the rest of the data gets thrown into the db without being specified |
| 17:29.39 | FlatFoot | ie dst , src , dstchannel |
| 17:29.40 | Corydon76-vcch | MindTheGap: use the SHELL() function for that |
| 17:29.51 | bkruse | :D |
| 17:30.05 | Corydon76-vcch | FlatFoot: that would be because you're using the standard names |
| 17:30.37 | blitzrage | MindTheGap: ya, you need to use SHELL(). It's not in 1.4, but it backports from trunk with a warning, but worked for me (i.e. I just copied func_shell.c into the funcs directory and compiled) |
| 17:30.39 | FlatFoot | yep thats what i thought calldate was |
| 17:31.04 | Corydon76-vcch | No, the 3 standard date fields are start, answer, and end. |
| 17:31.41 | FlatFoot | so i need to change the layout of my table then , have the names changed in v 14.11 then |
| 17:31.59 | FlatFoot | * 1.4.11 |
| 17:32.05 | Corydon76-vcch | No, the names have not changed since before 1.0 was released |
| 17:32.20 | FlatFoot | ah ok |
| 17:32.43 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 17:33.45 | *** join/#asterisk callguy (n=callguy@pool-71-162-97-18.bstnma.east.verizon.net) |
| 17:34.15 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 17:35.12 | *** join/#asterisk Sapote (n=guillote@host32.200-117-222.telecom.net.ar) |
| 17:39.10 | BadBru | who used chanspy succesfull until now ? |
| 17:39.32 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 17:40.49 | [TK]D-Fender | BadBru: Pretty much everyone who's tried. What's the problem? |
| 17:43.01 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:43.57 | *** join/#asterisk kuto (n=kuto@bas1-toronto36-1279554555.dsl.bell.ca) |
| 17:44.09 | BadBru | i did it// |
| 17:44.12 | BadBru | it's working |
| 17:45.01 | BadBru | etxen => 779,1,Chanspy(SIP/400) (where 400 is number i wish to listen on) |
| 17:45.32 | BadBru | someone knows how i can stop beeping from begining and end of the call ? |
| 17:45.55 | [TK]D-Fender | BadBru: "show application chanspy" <- try reading the instructions. |
| 17:48.04 | BadBru | i must add |qb instead of (sip/400) put (sip/400|qb) ? |
| 17:48.12 | BadBru | q= don't play a beep at begining of spy |
| 17:48.13 | [TK]D-Fender | BadBru: Certainly makes sense. How about trying it now... |
| 17:49.09 | _x86_ | haha |
| 17:49.19 | _x86_ | TK never fails to amuse ;) |
| 17:51.21 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 17:51.33 | BadBru | if put |qb --- it takes aprox 3-5 sec until spying |
| 17:51.50 | BadBru | and sip call is inrrerupting |
| 17:52.00 | BadBru | if put only |q it's ok |
| 17:52.45 | BadBru | seems like -b option is not preety fast.. it's slow (-b = listen only bridged calls) |
| 17:53.56 | kand | Can somebody help me? My spa2102 sends a bye on answer if the cid is more than four digits. |
| 17:58.35 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 18:02.27 | *** join/#asterisk badcfe (i=christia@alltid.dritings.no) |
| 18:04.16 | *** join/#asterisk miraclew (i=miraclew@125.118.7.187) |
| 18:05.15 | *** part/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 18:08.53 | *** join/#asterisk puga (n=none@189.5.213.166) |
| 18:09.22 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:19.05 | puga | anyone here can help with asterisk 1.4 CDR? |
| 18:19.41 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:20.42 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:25.19 | MindTheGap | hello guys, how do I reload say.conf |
| 18:28.30 | mikecx | module reload app_playback.so |
| 18:28.38 | mikecx | that will reload say.conf |
| 18:29.52 | MindTheGap | thanks mikecx |
| 18:29.54 | mikecx | is there a way to make a dialtone when someone presses X to get an outside line? A.k.a user presses 9, waits till dialtone, then dials |
| 18:30.01 | mikecx | MindTheGap: no prob |
| 18:33.42 | puga | mikecx if you get this answer, pass it to me please |
| 18:35.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:35.46 | kand | mikecx: How about an extension 9 that uses DISA with no password. |
| 18:36.07 | *** join/#asterisk fadey (n=fadey@84.76.46.249) |
| 18:36.36 | mikecx | kand: using DISA pretty easy? Haven't messed around with it yet? |
| 18:36.54 | kand | mikecx: very, just protect it from outside callers. |
| 18:37.28 | Qwell | or just add it to your dialplan... |
| 18:37.30 | kand | mikecx: But if you are doing this to keep an old standard why not just let them dial 9 infront of the number then cut it off |
| 18:37.32 | Qwell | what type of phones? |
| 18:37.47 | mikecx | Qwell: linksys 942's |
| 18:37.54 | Qwell | add it to your phones dialplan |
| 18:38.44 | puga | this old standard should die =P |
| 18:38.51 | Qwell | what standard? |
| 18:38.56 | mikecx | kand: just wanted to keep it like the old system |
| 18:39.04 | kand | understandable |
| 18:40.03 | puga | its good when you have more than one outside |
| 18:40.22 | kand | mikecx: I had the same request and made the system reconize both so the client could transition. |
| 18:40.52 | puga | kand you used DISA? |
| 18:41.29 | kand | kand: that is only way to recreate the dial a number then wait for a dial tone. The other is like qwell said and is more elegant. |
| 18:42.09 | *** part/#asterisk twitchnln (n=twitch@cpe-orncorp.dktc.atl.oneringnetworks.net) |
| 18:42.09 | mikecx | alright, I think i'll avoid that and tell them to piss off if they want that do to the security risks |
| 18:42.13 | kand | Or you can pattern match like exten => _9XXXXX.,1,Dial(SIP/${EXTEN:1}) |
| 18:42.24 | exothermc | Does someone have a good understanding what settings can go under specific contexts vs general context in voicemail.conf? specifically the imapserver setting? |
| 18:42.33 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 18:42.34 | kand | Then you can also do a pattern with no 9 |
| 18:42.38 | exothermc | Can you mix voicemail storage on the same instance of asterisk? |
| 18:42.44 | Qwell | exothermc: no |
| 18:43.01 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 18:43.25 | exothermc | Ok so there is one that doesn't cross down to a local context, any others? |
| 18:45.01 | exothermc | ok nm I just reread the voicemail.conf docs out at the voip-info site |
| 18:45.07 | exothermc | looks like it is pretty clear there. |
| 18:45.18 | Qwell | clear docs? at voip-info? |
| 18:45.22 | Qwell | They're probably wrong then |
| 18:45.23 | exothermc | Although no one seems to have added the imap setting. |
| 18:45.47 | exothermc | Qwell: Is there a better source? |
| 18:45.51 | Qwell | ~book |
| 18:45.52 | 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 |
| 18:45.58 | Qwell | or the config samples in the source |
| 18:47.50 | *** join/#asterisk MohShami (n=mohshami@86.108.45.49) |
| 18:48.08 | DarKnesS_WolF | EuroIAX seems cool |
| 18:48.34 | *** join/#asterisk fadey (n=fadey@84.76.46.249) |
| 18:48.57 | MohShami | hey guys, I created an H323 trunk between an ericsson PBX and an asterisk PBX, calls pass through normally between the PBXs, but when someone one the ericsson PBX tries to call a meetme conference on asterisk, a voice just says goodbye and hangs up, any ideas? |
| 18:50.18 | DarKnesS_WolF | MohShami: debug messages please ? |
| 18:50.21 | DarKnesS_WolF | on the asterisk |
| 18:50.22 | DarKnesS_WolF | side |
| 18:50.38 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 18:51.48 | MohShami | sadly I was kicked out of the office because they wanted to close up ^_^ |
| 18:51.48 | MohShami | I didn't see anything out of the ordinary in the logs |
| 18:51.48 | DarKnesS_WolF | MohShami: most7eel ;-) |
| 18:51.52 | MohShami | :D |
| 18:51.58 | MohShami | you speak arabeezee :D |
| 18:52.13 | DarKnesS_WolF | egyptian |
| 18:52.18 | DarKnesS_WolF | u ? |
| 18:52.18 | Kobaz | MohShami: do you have h323 peers working? |
| 18:53.07 | MohShami | DarkNesS_Wolf: Jordanian |
| 18:53.08 | DarKnesS_WolF | MohShami: ok did u try to have an IP user or a client from asterisk side to reach this meetme rom ? |
| 18:53.32 | Qwell | most7eel? words don't have numbers! |
| 18:53.34 | MohShami | if a user on the asterisk box called the conference everything goes fine |
| 18:53.41 | MohShami | Qwell |
| 18:53.59 | Kobaz | MohShami: poke |
| 18:54.00 | MohShami | some letters in arabic have no english representation, so we use numbers for them |
| 18:54.03 | Qwell | ahh |
| 18:54.06 | Qwell | MohShami: fair enough |
| 18:54.27 | Qwell | so what letter is 7? |
| 18:54.32 | MohShami | Kobaz: To be honest, I'm very new and I used trixbox for this |
| 18:54.37 | Kobaz | MohShami: ah |
| 18:54.43 | Kobaz | i'm trying to get h323 peers working |
| 18:54.44 | MohShami | Qwell: It's a stronger h |
| 18:54.51 | DarKnesS_WolF | Qwell: something like h bot u know ........ like mohamed |
| 18:54.53 | Qwell | MohShami: type it here? IRC should support it fine |
| 18:54.54 | Kobaz | i have the oooh323 channel driver working (it seems) |
| 18:55.17 | DarKnesS_WolF | Qwell: i can teach u arabic and u teach me more asterisk :P? |
| 18:55.25 | MohShami | Ø |
| 18:55.32 | Qwell | MohShami: ahh |
| 18:55.35 | DarKnesS_WolF | MohShami: ok u just call fro mteh PBX and u get goodboy ? |
| 18:55.38 | DarKnesS_WolF | bye * |
| 18:55.57 | Kobaz | anyone have any good docs on getting h323 peers working? ie: netmeeting using asterisk to call something else... ie: iax/sip |
| 18:56.01 | DarKnesS_WolF | seems ur dilaing a wrong number there is not goodbye message in teh MeetME application . |
| 18:56.25 | MohShami | when I call from ericsson to conference on asterisk I get goodbye, other than that it works fine |
| 18:56.25 | DarKnesS_WolF | Kobaz: why H323 ? |
| 18:56.30 | Kobaz | DarKnesS_WolF: because i have some phones that only do h323 and i want those to work as well |
| 18:56.32 | MohShami | some PBXs only support H323 |
| 18:56.46 | Kobaz | DarKnesS_WolF: and i'm using netmeeting for testing |
| 18:56.47 | MohShami | that's the same reason we're using it |
| 18:57.02 | MohShami | Kobaz: are you using an asterisk based distro? |
| 18:57.05 | Kobaz | no |
| 18:57.22 | Kobaz | i dont see the need |
| 18:57.57 | MohShami | I'm planing to build an asterisk box using freebsd |
| 18:58.07 | MohShami | I used trixbox as a PoC |
| 18:58.54 | Kobaz | ah |
| 18:59.39 | Sapote | :) |
| 18:59.41 | DarKnesS_WolF | MohShami: why FreeBSD ? |
| 18:59.55 | MohShami | I've been a Linux zealot for 3 years now |
| 19:00.02 | DarKnesS_WolF | MohShami: use a normal linux distro ... use something like debian if u want it stable |
| 19:00.05 | DarKnesS_WolF | zealot ? |
| 19:00.07 | Sapote | sound gooooood :) |
| 19:00.14 | MohShami | :D |
| 19:00.30 | MohShami | I got introduced to freebsd 2 weeks ago, for someone who learned linux using gentoo, it fealt right at home |
| 19:00.37 | MohShami | I fell in love instantly |
| 19:00.41 | DarKnesS_WolF | anyone used EuroIAX ? or has any better terminiation "A-Z" cheap / and accept CC or VISA no need for paypal or moneybokers ? |
| 19:01.33 | DarKnesS_WolF | MohShami: FreeBSD is okay .. i love OpenBSD still kicking a little bit wiht it ... but i don't see the need .. i already compiling my own asterisk i have some boxes using debian / ubuntu / mandriva and i don't see any diffrentce ;-) |
| 19:02.01 | [TK]D-Fender | If you're planning on using BSD, if you need Zaptel you're jsut asking for greif |
| 19:02.08 | MohShami | DarKnesS_WolF: all UNIX flavored OSs are fine, it's a matter of preference |
| 19:02.49 | MohShami | [TK]D-Fender: We only want the box to host MeetMe conference with calls forwarded from another PBX using H323, would that be a problem? |
| 19:02.52 | DarKnesS_WolF | i still have my signed zaptel card from mark ;-) |
| 19:03.05 | [TK]D-Fender | MohShami: Yes, as MeetMe requires Zaptel. |
| 19:03.16 | DarKnesS_WolF | i have the singuture on a paper but lazy to stick it to the card ... man it's been like a year since i meet him :( |
| 19:03.24 | [TK]D-Fender | MohShami: And H.323 is ANOTHER problem all its own... |
| 19:03.40 | DarKnesS_WolF | MohShami: yes meetme still needs a timing source if u don't have zaptel u can just modprobe ztdummy |
| 19:03.56 | [TK]D-Fender | DarKnesS_WolF: ... ZTDUMMY *is* Zaptel |
| 19:03.57 | DarKnesS_WolF | [TK]D-Fender: how are u doing dude :-) |
| 19:04.22 | MohShami | hmm |
| 19:04.33 | MohShami | so I should just go with CentOS? |
| 19:05.20 | [TK]D-Fender | MohShami: Good choice, well documented |
| 19:05.23 | DarKnesS_WolF | MohShami: i never used CentOS but it seems good all asterisk guys like it .. i love debian ... |
| 19:05.41 | DarKnesS_WolF | [TK]D-Fender: ah u were talking about the moddule ... i thought u ment the card :-) |
| 19:05.45 | DarKnesS_WolF | my bad |
| 19:06.06 | MohShami | DarKnesS: again, it's a matter of preference, I prefer Redhat based distros for me servers |
| 19:06.25 | MohShami | sot ZTDUMMY won't work under freebsd? |
| 19:06.47 | MohShami | I got H323 working on a PC with no zaptel cards, does that have anything to do with it? |
| 19:08.06 | DarKnesS_WolF | MohShami: no h323 should work normal. |
| 19:08.33 | DarKnesS_WolF | what i think that it is issue with ur dialplan .. cuz meetme application shouldn ot have the word " GoodBye" |
| 19:08.38 | tzafrir | For some definition of "normal" |
| 19:08.59 | MohShami | the test system is running centos |
| 19:09.29 | MohShami | usually I would learn how to use something from the command line and configuration files, no gui, but I needed to do this one quickly |
| 19:10.03 | DarKnesS_WolF | MohShami: ahh u did use GUI ? trixbox or what ? |
| 19:11.04 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 19:11.19 | DarKnesS_WolF | MohShami: now i'm sure it is a dialplan issue |
| 19:11.54 | *** join/#asterisk oskude (n=oskude@dslb-084-058-230-121.pools.arcor-ip.net) |
| 19:16.18 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:21.15 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 19:22.02 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 19:22.03 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 19:23.01 | funxion | anyone have a clue why on an E&M t1 both in and outbound calls disconnect right after answer? |
| 19:23.02 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:26.40 | *** part/#asterisk myiagy (n=Jose@200.215.59.133) |
| 19:27.57 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 19:31.19 | funxion | anyone here |
| 19:32.26 | DarKnesS_WolF | funxion: never used E&M T1 |
| 19:32.29 | DarKnesS_WolF | may be codec ? |
| 19:32.34 | funxion | no |
| 19:32.38 | DarKnesS_WolF | what is the error message on teh asteriks CLI ? |
| 19:32.41 | funxion | I can hear a split second of audio |
| 19:32.47 | funxion | I think its a polarity issue |
| 19:32.54 | funxion | but need confimation on that |
| 19:32.54 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 19:33.27 | *** join/#asterisk hohum_ (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 19:33.32 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 19:37.28 | DarKnesS_WolF | funxion: what u think ? what do u mean ? |
| 19:40.25 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:40.46 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 19:44.22 | tzafrir | Is there a way in the asterisk CLI to check to which groups a zap channel belongs? |
| 19:46.53 | DarKnesS_WolF | tzafrir: dont think so |
| 19:48.47 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 19:49.03 | funxion | tzafrir why? |
| 19:49.13 | funxion | dont you have access to zapata.conf? |
| 19:49.37 | tzafrir | gee. But what if I just edited something and have no idea if it was applied? |
| 19:49.57 | funxion | o |
| 19:49.59 | funxion | lol |
| 19:50.20 | tzafrir | funxion, and there's also users.conf |
| 19:50.21 | funxion | zap show channel X doesnt give group ionfo |
| 19:51.03 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 19:51.06 | tzafrir | if you have one thing in [channels] of zapata.conf and another in [general] of users.conf, what sets the default to the sections of users.conf? |
| 19:51.11 | tzafrir | Do you actually remember? |
| 19:51.47 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:53.04 | funxion | nope |
| 19:54.53 | DarKnesS_WolF | Gtalk + aserisk sweeet ! |
| 19:55.04 | DarKnesS_WolF | if they just can support a little bit of a small bandwitdh codec it will be amazing |
| 19:55.37 | galeras | Dear sirs, i have many calls ended by agent with 0 seconds of calltime, does this mean the agent has finished the call as soon as he receive that? (pls take a look off my queue_log at http://www.pastebin.ca/815097) |
| 19:58.16 | [TK]D-Fender | galeras: Means you've got an agent who's hangining up on callers |
| 19:58.25 | tzafrir | Is there any decent IM client that can show status of extensions in Asterisk? |
| 19:58.40 | [TK]D-Fender | galeras: WE've had one of those who we busted... for the longest time they thought it was our system dropping calls |
| 19:58.57 | [TK]D-Fender | tzafrir : eyeBeam can. |
| 19:59.15 | [TK]D-Fender | tzafrir : And you called them... EXTENSIONS :( |
| 19:59.17 | tzafrir | So can twinkle. But it is not really an IM |
| 19:59.24 | [TK]D-Fender | tzafrir : I'd use a web-app personally. |
| 19:59.49 | tzafrir | [TK]D-Fender, I need something for my computer. Not a clumsy web app |
| 20:01.14 | tzafrir | a plugin for gaim would have been great for me |
| 20:01.15 | *** join/#asterisk tw-nym (n=tw@peer.chuui.jp) |
| 20:01.22 | [TK]D-Fender | tzafrir : Pidgin supposedly does SIP IM, maybe supports presence.. |
| 20:01.46 | lirakis | [TK]D-Fender: pidgin does sip?? .. |
| 20:02.07 | tzafrir | Just tried it (my version is still called gaim). It was practically useless as far as presense is conrecened |
| 20:02.16 | lirakis | [TK]D-Fender: oh wow.. yeah it does.. i see it right here |
| 20:02.36 | tzafrir | I did not get a list of "buddies" from the server, and I couldn't even subscribe a new one |
| 20:03.08 | tzafrir | twinkle 1.1 uses publish/subscribe. And this generally works |
| 20:03.40 | funxion | anyone have a clue why on an E&M t1 both in and outbound calls disconnect right after answer? Could it be a polarity issue? |
| 20:03.48 | tzafrir | lirakis, only SIMPLE (SIP instant messaging) |
| 20:04.29 | lirakis | tzafrir: yeah i see that |
| 20:04.54 | [TK]D-Fender | funxion: consider "E&M" vn "E&M Wink" |
| 20:04.57 | [TK]D-Fender | vs* |
| 20:08.45 | galeras | <PROTECTED> |
| 20:08.53 | *** join/#asterisk newbie289 (n=harry@122.164.198.182) |
| 20:09.10 | funxion | TK sry it is e&m wink |
| 20:09.12 | [TK]D-Fender | galeras: Yeah... install a camera and watch that bastard like a hawk. |
| 20:09.16 | funxion | I forgot to specify |
| 20:09.37 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 20:10.18 | galeras | <PROTECTED> |
| 20:10.31 | [TK]D-Fender | galeras: :) |
| 20:10.56 | km- | wow, d-fender caused someone to get fired? :P |
| 20:11.10 | km- | [TK]D-Fender: that's definitely a ribbon to add to your fruit salad. |
| 20:11.15 | funxion | lol |
| 20:11.17 | [TK]D-Fender | km-: No, I'm nt that fast... his HR is however though likely for similar reasons. |
| 20:11.35 | galeras | Do not worry he plead guilty himself |
| 20:12.08 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:12.54 | km- | hehe, I wonder how often it happens where I work. |
| 20:14.22 | *** join/#asterisk BadBru (n=bad_b@ACA2A975.ipt.aol.com) |
| 20:14.22 | exothermc | is it possible to manage VM contexts with realtime? |
| 20:16.08 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 20:16.32 | mikecx | [TK]D-Fender: think you could help me with failover/multiple lines now that I have access to my conf files? |
| 20:16.44 | *** join/#asterisk dty (n=dertybiz@195.225.54.221) |
| 20:18.40 | *** join/#asterisk mog (i=mog@nat/digium/x-5b2078bbbb11d46c) |
| 20:18.40 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:19.27 | [TK]D-Fender | mikecx: like I told you before just set "group=[0-31]", and make sure you are dialing through the group : Dial(Zap/g[0-31]/12517235533) |
| 20:20.24 | tzafrir | group [0-31]? |
| 20:20.30 | galeras | Sirs, Any bad experience with redfone's fonebridge2-EC?, Do you advice to use it? |
| 20:21.25 | mikecx | [TK]D-Fender: does group have to be set individually? |
| 20:21.48 | tzafrir | [TK]D-Fender, typedef unsigned long long ast_group_t: there are 64 groups |
| 20:21.54 | [TK]D-Fender | mikecx: You can, but don't have to. if you multiple channels at once, they all take the same group |
| 20:22.11 | [TK]D-Fender | tzafrir : IIRC there are only 32 possible groupings |
| 20:22.25 | [TK]D-Fender | tzafrir : Ok, fine, sure |
| 20:24.48 | mikecx | [TK]D-Fender: what's the third parameter of Dial()? |
| 20:25.17 | [TK]D-Fender | mikecx: "show application dial" |
| 20:26.59 | mikecx | [TK]D-Fender: so in essense, not needwed persay if you don't need to configure? |
| 20:29.40 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:30.33 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 20:31.31 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:33.34 | *** join/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 20:34.19 | BadBru | someone know why i get this error |
| 20:34.20 | BadBru | Failed to execute '/usr/share/asterisk/agi-bin/cidspoof.agi': No such file or directory |
| 20:34.20 | BadBru | <PROTECTED> |
| 20:34.20 | BadBru | <PROTECTED> |
| 20:34.32 | BadBru | Auto fallthrough, channel ... |
| 20:34.51 | admgecko | hello everyone, ive got a really strange problem, ive just compile asterisk from svn on debain etch |
| 20:35.27 | admgecko | and finally got an fxo / fxs that works (spa3102) and i can now call out using it but... |
| 20:36.14 | admgecko | my two extentions cant call each other, it goes stright to voicemail...the console says |
| 20:36.23 | admgecko | Everyone is busy/congested at this time (1:0/0/1) |
| 20:36.59 | admgecko | but i can call outbound to the world via the pstn / sipgate / fwd....and i can dial other extention via a group |
| 20:37.52 | admgecko | both extentions are in the same context, but 777 for simulate incoming call rings engaged |
| 20:38.06 | km- | admgecko: the spa3102 has two lines? Did you configure them with individual sip accounts? |
| 20:38.45 | admgecko | km: yeah, the phone is on the fxs, as extention 103, and the line in set to route to the asterisk box |
| 20:39.01 | km- | so you're trying to call yourself from the same extension? |
| 20:39.21 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:39.24 | admgecko | km: no im trying to call extention 102, my sip hardphone |
| 20:39.24 | km- | i.e., you pick up the FXS phone, and dial a context that simulates an incoming call on the FXO, thereby bridging back to the FXS? |
| 20:39.29 | km- | ah. |
| 20:39.45 | km- | does asterisk report anything about codec oddities? |
| 20:39.56 | admgecko | not that i can see |
| 20:40.17 | admgecko | i can call outbound via the pstn by dialing 9 |
| 20:40.28 | BadBru | some1 know about this error |
| 20:40.29 | BadBru | <PROTECTED> |
| 20:40.29 | BadBru | ? |
| 20:40.30 | km- | from both the hardphone and the FXS port? |
| 20:40.42 | admgecko | yup |
| 20:41.10 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 20:41.13 | km- | BadBru: you can ignore it, it means the call is done but asterisk wasn't sure what status it was left... |
| 20:41.16 | nestAr | hate nickserv |
| 20:41.19 | nestAr | H8 |
| 20:41.52 | km- | admgecko: can you call hardphone->FXS or FXS->hardphone, or are both directions doing the same thing? |
| 20:42.14 | [TK]D-Fender | BadBru: Means your exten has not more priorities to execute and the call is ending |
| 20:42.22 | admgecko | both directions are doing the same thing |
| 20:42.36 | BadBru | aha then the main problem is |
| 20:42.37 | BadBru | Failed to execute '/usr/share/asterisk/agi-bin/cidspoof.agi': No such file or directory |
| 20:42.37 | BadBru | <PROTECTED> |
| 20:42.50 | BadBru | <PROTECTED> |
| 20:42.55 | km- | BadBru: that would do it. |
| 20:43.03 | BadBru | it exists and have run permission |
| 20:43.09 | admgecko | ringing straight to voicemail, and if i take the vm off, the ring engaged |
| 20:43.25 | km- | BadBru: is it executable by everyone or just the user that created it? |
| 20:43.27 | [TK]D-Fender | BadBru: maybe it exists and has permissions, but is not EXECUTABLE |
| 20:43.31 | admgecko | but if i put them in a group and call the group, they ring |
| 20:43.57 | admgecko | but 7777 for simulate incoming call rings enageded |
| 20:44.18 | BadBru | [TK]D-Fender, chmod +x cidspoof.agi isn't enough ? |
| 20:44.20 | admgecko | even though it just points to the 600 group |
| 20:44.22 | km- | admgecko: I almost want to believe it's a codec problem but I have no concrete evidence to prove it. Try forcing both sip phones to g711u by putting "disallow=all" followed by "allow=ulaw" in the sip configs and restart.. Try it again. |
| 20:44.40 | km- | BadBru: execute ls -l /usr/share/asterisk/agi-bin/cidspoof.agi |
| 20:44.43 | admgecko | right, hold on a second |
| 20:44.50 | km- | BadBru: and paste it. |
| 20:46.47 | BadBru | -rwxr-xr-x 1 root root 2706 Dec 13 15:45 /usr/share/asterisk/agi-bin/cidspoof.agi |
| 20:47.28 | km- | interesting |
| 20:47.32 | BadBru | i have another agi script and have same permissions |
| 20:47.39 | BadBru | and it's working |
| 20:47.49 | BadBru | i think is somthing from cidspoof.agi |
| 20:47.53 | BadBru | inside |
| 20:48.12 | km- | does cidspoof.agi have the first hashbang that explains what interpreter to use? |
| 20:48.26 | km- | though I'd believe more that that would yield a bad interpreter/permission denied error |
| 20:48.44 | BadBru | #!/usr/bin/perl |
| 20:48.45 | BadBru | $|=1; |
| 20:48.45 | BadBru | while(<STDIN>) { |
| 20:48.45 | BadBru | <PROTECTED> |
| 20:48.45 | BadBru | <PROTECTED> |
| 20:48.45 | BadBru | <PROTECTED> |
| 20:48.47 | BadBru | <PROTECTED> |
| 20:48.49 | BadBru | <PROTECTED> |
| 20:48.51 | BadBru | } |
| 20:48.55 | BadBru | sub checkresult { |
| 20:48.56 | BadBru | <PROTECTED> |
| 20:48.57 | BadBru | <PROTECTED> |
| 20:48.57 | [TK]D-Fender | BadBru: Do not spam in here |
| 20:48.59 | BadBru | <PROTECTED> |
| 20:48.59 | km- | wtf |
| 20:49.01 | BadBru | <PROTECTED> |
| 20:49.03 | BadBru | <PROTECTED> |
| 20:49.03 | km- | I didnt ask for the whole file :) |
| 20:49.05 | BadBru | <PROTECTED> |
| 20:49.07 | BadBru | <PROTECTED> |
| 20:49.09 | BadBru | <PROTECTED> |
| 20:49.11 | BadBru | ok sorry :) |
| 20:49.27 | BadBru | i think it's going to else |
| 20:49.53 | km- | something's awry with asterisk trying to find the file in the first place |
| 20:49.56 | km- | what, I'm not too sure. |
| 20:50.17 | admgecko | km: added that to the phones defination in /etc/asterisk/sip_addtional.conf |
| 20:50.20 | [TK]D-Fender | BadBru: No, its NOT, you can see all to clear that : Failed to execute '/usr/share/asterisk/agi-bin/cidspoof.agi': No such file or directory <-- your FILE is no good, forget the CONTENTS |
| 20:50.37 | admgecko | reloaded asterisk / rebooted phone / rebooted spa3102 |
| 20:50.40 | admgecko | same thing :( |
| 20:50.57 | BadBru | [TK]D-Fender, what do you mean my file is not good ? |
| 20:51.20 | admgecko | km: i can pm the console output if you would like |
| 20:51.27 | BadBru | if i put to the same fine.. chmod -x then it say.. doesn't have permission to execute |
| 20:51.33 | km- | admgecko: why dont you pop it in a pastebin and shoot me the link |
| 20:51.34 | BadBru | then file exists |
| 20:51.49 | admgecko | hold on, ill bang it on the webserver ;) |
| 20:52.36 | [TK]D-Fender | BadBru: it means for about the INSIDE of your file, its the attributes & location taht are bad |
| 20:53.09 | [TK]D-Fender | BadBru: What user is * running as? |
| 20:53.19 | BadBru | root |
| 20:53.43 | admgecko | km: can i pm link? |
| 20:53.46 | km- | admgecko: are you running really verbose? i.e., asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvgc |
| 20:53.47 | km- | sure. |
| 20:53.58 | admgecko | no but i can be.... |
| 20:54.19 | km- | yeah, lets kick the verbosity up. |
| 20:54.22 | [TK]D-Fender | BadBru: Go try and execut it manually. |
| 20:54.38 | BadBru | something strange.. same cidspoof.agi works on older asterisk |
| 20:55.18 | [TK]D-Fender | BadBru: Don't waste time with comments like that. Those were different circumstances and do not apply to NOW. |
| 20:55.33 | [TK]D-Fender | BadBru: Try to run it at CLI yourself now. |
| 20:56.29 | funxion | does anyone have any experience with France Telecom? |
| 20:56.36 | funxion | Ds0's specifically |
| 20:56.45 | BadBru | sorry for lazy question.. how i run agi script from cli ? |
| 20:57.43 | km- | just try running it. |
| 20:57.53 | km- | type the filename in and hit enter, see what happens |
| 20:58.05 | *** part/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 20:58.06 | [TK]D-Fender | like this : /usr/share/asterisk/agi-bin/cidspoof.agi |
| 20:58.18 | km- | hmmm |
| 20:58.20 | [TK]D-Fender | km-: ... lol :) |
| 20:58.23 | *** join/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 20:58.46 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:58.54 | km- | [TK]D-Fender: didn't realize we were running remedial linux use 101 here. ;) |
| 20:59.02 | BadBru | *CLI> /usr/share/asterisk/agi-bin/cidspoof.agi |
| 20:59.02 | BadBru | No such command '/usr/share/asterisk/agi-bin/cidspoof.agi' (type 'help' for help) |
| 20:59.08 | BadBru | hmm |
| 20:59.32 | [TK]D-Fender | BadBru: Well, despite all your claims that everything is right, its pretty clearly NOT. |
| 20:59.58 | BadBru | what's the problem ? |
| 21:00.13 | km- | well, for one thing, try running it on the linux command line, not the asterisk command line :) |
| 21:00.15 | admgecko | the strange this is that outbound is fine.... |
| 21:00.30 | km- | admgecko: can you put extensions.conf up on your site too? |
| 21:00.33 | admgecko | and it works via 600 (the group ring) |
| 21:00.40 | admgecko | yeah np hold on |
| 21:01.03 | *** part/#asterisk galeras (n=galeras@200.31.204.42) |
| 21:01.49 | admgecko | im using freepbx so do you want extentions_additional.conf as well? |
| 21:01.56 | Qwell | ~freepbx |
| 21:02.08 | 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 |
| 21:02.32 | km- | eh, yeah, if you're using freepbx I dont know how much help I can be |
| 21:03.07 | admgecko | is that a dump the bloody gui and run make samples again, and write my own config's cos this is just making a mess of things? |
| 21:03.09 | admgecko | ;) |
| 21:03.18 | Qwell | admgecko: yes |
| 21:03.47 | Nugget | jbot's way sounds nicer. :) |
| 21:03.53 | admgecko | might need some help then ;) anyone fancy walking me though asterisk setup 101 ;) |
| 21:03.59 | Qwell | ~book |
| 21:03.59 | 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 |
| 21:04.16 | km- | yeah, download that pdf and get reading :) |
| 21:04.33 | admgecko | grand, im sure ill have some question for you soon :D |
| 21:04.41 | [TK]D-Fender | admgecko: Here a leg-up for you : |
| 21:04.43 | km- | as long as you're using asterisk, we can try to help :) |
| 21:04.45 | [TK]D-Fender | ~jerjerguide |
| 21:04.45 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 21:04.51 | km- | oh nice |
| 21:04.54 | km- | jerjer wrote a quickstart? |
| 21:04.59 | km- | that'll be useful. |
| 21:05.02 | km- | jerjer is a smart guy |
| 21:05.21 | [TK]D-Fender | km-: And had some help validaing and tweaking it ;) |
| 21:05.48 | admgecko | to dump freepbx, can i just do an rm * in the /etc/asterisk dir and run make samples in the asterisk src directory? |
| 21:05.59 | admgecko | then delete amportal.conf? |
| 21:06.04 | admgecko | and the webfiles? |
| 21:06.06 | Qwell | admgecko: get rid of the php too |
| 21:06.14 | admgecko | grand |
| 21:06.18 | Qwell | and, uninstall php and apache from your system, so you don't get tempted to install it again :P |
| 21:06.35 | mikecx | when I dial my slatrunk all the internal phones ring. Any ideas? |
| 21:06.53 | admgecko | cant do that, its running subversion + horde webmail :) |
| 21:06.53 | [TK]D-Fender | Qwell: Don't forget to tell him to salt the earth! |
| 21:08.33 | admgecko | thanks for all the help guys, i know it can be dull helping out the n00bs like me :-p |
| 21:09.16 | km- | we hang out here purely for the enjoyment thereof. :) |
| 21:09.42 | admgecko | hey are any of you guys using Avaya 4610Sw phones? |
| 21:10.27 | [hC] | [TK]D-Fender: i didnt know you were in canada? |
| 21:11.38 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-121-28.pskn.east.verizon.net) |
| 21:12.42 | [TK]D-Fender | [hC]: I'm sure you did... |
| 21:13.09 | [hC] | huh... if i did i forgot.. :) |
| 21:13.54 | admgecko | should be running make samples, or just create the config files i need? |
| 21:14.30 | km- | make samples will help you understand the verbage and syntax of the conf files. |
| 21:15.02 | BadBru | who know what can pe wrong in that agi script |
| 21:15.03 | BadBru | http://www.rootsecure.net/content/temp/cidspoof.agi |
| 21:15.31 | [TK]D-Fender | BadBru: You are not listening.. its isn't the CONTENTS! Your permissions or folder structure is bad. |
| 21:15.37 | BadBru | as i get No such file or directory when asterisk executes it |
| 21:17.23 | BadBru | [TK]D-Fender permission is not problem i have another agi script in same folder wich executes without any problem |
| 21:17.42 | BadBru | ls -al show them with same permission atribute |
| 21:18.10 | [TK]D-Fender | BadBru: You can't even call it from Linux CLI. its not the contents.... |
| 21:18.43 | km- | d-fender |
| 21:18.51 | km- | you misread his paste |
| 21:18.55 | km- | he tried to run it from the asterisk cli |
| 21:18.57 | km- | not the linux cli |
| 21:19.00 | km- | I set him straight, though. |
| 21:19.20 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 21:19.39 | BadBru | cli says No such command '/usr/ bla bla |
| 21:19.51 | BadBru | this is the problem ? |
| 21:20.01 | De_Mon | [Dec 13 16:22:28] WARNING[8653]: chan_iax2.c:797 jb_warning_output: Resyncing the jb. last_delay 1, this delay 65537, threshold 1008, new offset -65537 |
| 21:20.59 | km- | De_Mon: latency issues between your iax2 endpoint and asterisk? Something wonky with your jitter buffer. |
| 21:21.46 | km- | did you just adjust the time of your asterisk box maybe? |
| 21:21.49 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 21:22.53 | BadBru | time is ok |
| 21:22.58 | admgecko | in the sip.conf; do device specific settings override general settings |
| 21:23.08 | [TK]D-Fender | km-: I DID have him run it from LINUX CLI. |
| 21:23.26 | admgecko | ie if i have port=5060 in general, and port=5061 for some phones, will the phones use the right port? |
| 21:23.31 | BadBru | date 13th.. maybe this is the reason :) |
| 21:23.35 | [TK]D-Fender | km-: and... I now see that he has the attention span of a dust mite... |
| 21:23.58 | km- | [TK]D-Fender: Look! a bunny! heh. |
| 21:24.12 | km- | admgecko: I believe so, yes. |
| 21:24.24 | admgecko | cool cheers :D |
| 21:24.55 | mikecx | anyone help me setup sla? |
| 21:24.56 | km- | np |
| 21:25.47 | admgecko | well while we're on the subject....:-p |
| 21:26.08 | BadBru | [TK]D-Fender what u suggest me to try ? |
| 21:26.13 | [TK]D-Fender | mikecx: very few people use it OR the GUI... you are compounding the odds against you... |
| 21:26.42 | mikecx | [TK]D-Fender: stopped using the gui |
| 21:27.09 | ManxPower | mikecx: you can't stop using the GUI. You have to remove Asterisk and install the non-gui version |
| 21:27.20 | ManxPower | (or at least the asterisk config files) |
| 21:27.36 | [TK]D-Fender | mikecx: go read the docs folder for info on how to use SLA. |
| 21:28.57 | BadBru | [TK]D-Fender i found somewhere this error No such file or directory could mean.. bad interpereter |
| 21:29.59 | De_Mon | km- the time does look a bit off, but it wasnt changed during the call or anything like that |
| 21:32.13 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 21:37.33 | lirakis | later everyone |
| 21:37.40 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:38.00 | *** join/#asterisk lazybrain (n=kvirc@24.229.241.237.res-cmts.sm.ptd.net) |
| 21:38.07 | admgecko | ok so ive created a sip.conf and extensions.conf and voicemail.conf, just rebooting the phones :D |
| 21:38.24 | *** part/#asterisk ManxPower (n=manxpowe@246.sub-70-221-77.myvzw.com) |
| 21:38.42 | lazybrain | I want to be able to transfer a callering directory to someones voicemail without it ringing (I.E. you dont want to speak with them) can someone point me in the right direction? |
| 21:41.01 | Greek-Boy | how many hex characters are 27 bits? |
| 21:42.00 | admgecko | errmm 7 |
| 21:42.11 | Greek-Boy | lol |
| 21:42.12 | Greek-Boy | k |
| 21:42.13 | Greek-Boy | thanks |
| 21:42.26 | admgecko | you cant fit it in 6, cos that would be 24bits |
| 21:43.05 | lazybrain | anyone around ? |
| 21:43.25 | Greek-Boy | damn |
| 21:43.28 | admgecko | hex is alway 4 bits per letter, 2 hex digits per byte, to find the number of hex digits, just divide by 4, and always round up |
| 21:43.28 | Greek-Boy | I got a problem |
| 21:43.30 | BadBru | some1 here wich knows why an agi script is not running ? |
| 21:43.38 | admgecko | :d |
| 21:44.11 | Greek-Boy | my WIP330 phone only supports a key up to 27 bits but I'm using WAP and that needs a minimum of 32 bits? |
| 21:44.31 | lazybrain | can you not use wap ? |
| 21:44.45 | lazybrain | can you upgrade the phone firmware ? |
| 21:44.47 | admgecko | what kind of wap are you using, wep keys are 27/32 but |
| 21:44.51 | admgecko | *bit |
| 21:45.04 | admgecko | its to do with encoding |
| 21:45.26 | Greek-Boy | lemme try upgrading |
| 21:45.37 | admgecko | if you've lock the network down to 128bit wep (which is only midley better than no encyption at all) |
| 21:45.54 | admgecko | you will have a 27 charector key |
| 21:46.18 | admgecko | but you should really upgrade to WPA at a minuimum and WAP2 w/ mac filtering, if you can |
| 21:46.39 | admgecko | im using wpa2 w/ 802.1x radius auth, but im a geek :-p |
| 21:46.47 | lazybrain | im trying to setup an extension so I can forward a call directly to someones voicemail without the phone ringing. When I do this I get the VoicemailMain prompt |
| 21:46.57 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 21:48.47 | admgecko | lazybrain: i dont know much about asterisk, but you could create an application to go stright to their voicemail? |
| 21:48.47 | admgecko | like a ncfw on a merdian? |
| 21:49.27 | lazybrain | this has a to be easy, but I'm new to asterisk |
| 21:49.55 | admgecko | yeah me to, im a network / server / median guy |
| 21:50.13 | lazybrain | I tried this but it doesnt work. |
| 21:50.16 | lazybrain | exten => _*110,1,Voicemail(u${EXTEN:1}) ;send direct to VM |
| 21:51.25 | *** join/#asterisk adeeln (n=adeeln@c-24-7-132-155.hsd1.ca.comcast.net) |
| 21:51.59 | lazybrain | wheres a guru ? |
| 21:52.30 | adeeln | does anyone have any recommendations on a high availability / failover setup for asterisk they might be willing to share? |
| 21:52.42 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 21:54.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:56.13 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 21:57.24 | kand | lazybrain: that looks fine. What happens when you call it? |
| 21:58.43 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:59.16 | lazybrain | kand - one sec |
| 22:00.36 | lazybrain | kand - nothing happens |
| 22:00.42 | lazybrain | I press *110 and it does nothing |
| 22:01.04 | kand | lazybrain: do you see it on the cli? |
| 22:01.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:01.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:01.50 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 22:02.31 | lazybrain | kand - nope |
| 22:03.28 | kand | lazybrain: ok, it may be an issue with your phones dial plan and the *. What kind of phone? |
| 22:03.29 | *** join/#asterisk NovceGuru (i=shelby@ballmung.easymac.org) |
| 22:03.31 | lazybrain | kand - cisco7960 |
| 22:04.39 | NovceGuru | I'm playing with askoziaPBX on an embedded pc...pretty cool :D |
| 22:05.12 | kand | lazybrain: I dont think those have an issue by default. Try exten => *110,1,Voicemail(u${EXTEN:1}@context) where you replace context with appropriate one |
| 22:05.35 | lazybrain | kand - let me try it |
| 22:06.06 | blitzrage | Voicemail(exten@context,u) is the preferred format |
| 22:06.25 | kand | blitzrage: depending on asterisk version, I was taking it one step at a time |
| 22:06.32 | blitzrage | just an FYI :) |
| 22:06.42 | kand | blitzrage: good point ty |
| 22:06.48 | blitzrage | I always talk in 1.4ism's |
| 22:06.57 | blitzrage | 1.2 is dead to me! |
| 22:07.04 | kand | I hear ya |
| 22:07.06 | blitzrage | russellb: ^^^^ |
| 22:07.52 | lazybrain | lol |
| 22:08.02 | lazybrain | once I get it working, I want it to work for all extensions. |
| 22:08.47 | kand | lazybrain: exten => _*XXX,1,Voicemail(${EXTEN:1}@context,u) |
| 22:09.07 | lazybrain | kand trying now |
| 22:09.31 | kand | lazybrain: for 3 digit extension. Also you may want to test if the box exists and handle exceptions because who knows what users will type. |
| 22:10.40 | lazybrain | wierd The first way still doesnt work |
| 22:10.46 | lazybrain | I did extensions reload |
| 22:11.15 | kand | lazybrain: and you still dont see any activity in the cli when you dial the extension? |
| 22:11.25 | russellb | blitzrage: orly! |
| 22:11.40 | lazybrain | kand maybe im doing it wrong. I answer my phone, then I dial *110 and nothing happens. |
| 22:11.49 | blitzrage | russellb: yes sir -- all answers immediately assume you are running 1.4+! |
| 22:12.11 | kand | lazybrain: You would have to blind transfer to *110 |
| 22:12.24 | russellb | blitzrage: yay :) |
| 22:12.34 | blitzrage | ok... time to order a pizza |
| 22:12.44 | lazybrain | kand - k |
| 22:13.00 | kand | lazybrain: to work the way you where doing it read up on features.conf |
| 22:13.14 | kand | lazybrain: lot more work tho |
| 22:13.50 | lazybrain | I did a blind transfer and it hung up on the call |
| 22:14.47 | kand | lazybrain: try just calling *110 |
| 22:15.47 | lazybrain | kand it hangs |
| 22:16.29 | lazybrain | Incoming call: Got SIP response 404 "Not Found" |
| 22:16.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:16.39 | [TK]D-Fender | now is the the point where we realize just how far off the path we have gone and all the time lost. This is where we realize that LONG ago we should have asked for several things.... |
| 22:16.51 | [TK]D-Fender | the first is SIP DEBUG of the failed call attempt! |
| 22:17.02 | [TK]D-Fender | lazybrain, PASTEBIN IT. |
| 22:17.03 | [TK]D-Fender | ~pb |
| 22:17.04 | 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 |
| 22:17.05 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 22:17.08 | Greek-Boy | are there any wifi phones out there that support PTT? |
| 22:17.10 | kand | lazybrain: Dont work you are in good hands now.... |
| 22:17.13 | [TK]D-Fender | lazybrain, along with your DIALPLAN. |
| 22:18.47 | kand | I thought it was going to be to simple to ask for a pastebin, I guess live and learn |
| 22:20.18 | lazybrain | kand - I can pastebin that portion of the dial plan but I dont really want the rest for everyone to view. |
| 22:20.39 | kand | lazybrain: should be fine, as much as you can |
| 22:20.51 | [TK]D-Fender | kand, its NEVER simple, and you should NEVER trust a newb to have done anything right. Ask for all the goods and sort it out when it arrives. Save you hours of beating your head against a wall for a person to present you a useless tunnel-vision view of his problem. |
| 22:21.20 | kand | [TK]D-Fender: noted! ty |
| 22:21.36 | file | [TK]D-Fender: you have... an apprentice? |
| 22:21.36 | [TK]D-Fender | lazybrain, pastebin the relevant context(s), and while you're at it, your sip.conf [general] sectio, and your phone's section as well |
| 22:21.57 | lazybrain | ok |
| 22:22.41 | hmmhesays | what the heck autoconf.h: No such file or directory |
| 22:22.59 | outtolunc | lucky you <G> |
| 22:24.07 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) [NETSPLIT VICTIM] |
| 22:24.18 | blitzrage | yum install autoconf? :) |
| 22:25.32 | lazybrain | kand - http://pastebin.com/d2e225d61 |
| 22:25.38 | outtolunc | that or kernel-headers |
| 22:26.25 | [TK]D-Fender | lazybrain, I'm still missing the complete SIP debug for the failed attempt and I want to see the context header section up there as well |
| 22:26.47 | kand | lazybrain: umm, can you paste the cli |
| 22:26.48 | hmmhesays | yeah the guy who did this install doesn't know anything about linux |
| 22:26.57 | lazybrain | I got it working. THe context was wrong |
| 22:27.03 | [TK]D-Fender | ...... |
| 22:27.04 | kand | lol |
| 22:27.16 | lazybrain | tk d-fender how do you even get the complete sip debug ? sorry |
| 22:27.30 | [TK]D-Fender | lazybrain, "sip debug" at CLI |
| 22:28.03 | lazybrain | ok |
| 22:29.38 | [TK]D-Fender | kand, and never trust dialplan extens without seeing contexts specified.... |
| 22:30.01 | Greek-Boy | come on guys, I'm sure someone here knows about a good PTT solution that works well with asterisk. I have been tasked with replacing old radios with a IP PTT phone solution to integrate with asterisk... |
| 22:30.26 | kand | [TK]D-Fender: good point. didn't give us much to work with either.... |
| 22:31.36 | [TK]D-Fender | kand, And you should note every standard thing they are lacking and roast them IMMEDIATELY for it |
| 22:33.14 | kand | [TK]D-Fender: can do. thanks. Obviously new to this side of things. |
| 22:33.44 | lazybrain | wow, all this regarding my question |
| 22:33.54 | dexpdx | err I hate the dialplan format |
| 22:34.05 | kand | Ya I am new to answering question, just trying to give back... |
| 22:34.11 | [TK]D-Fender | lazybrain, what can I say, you're a common situation in here |
| 22:34.44 | lazybrain | TK d-fender - its cool, I'll look for you in a freebsd channel :) |
| 22:35.06 | [TK]D-Fender | dexpdx, http://go-cry-emo-kid.ytmnd.com/ |
| 22:35.44 | dexpdx | [TK]D-Fender: very funny haha |
| 22:37.03 | hmmhesays | this is the strangest install, now I can't modprobe ztdummy after I build it, it seems to be there |
| 22:39.11 | adeeln | which cisco phones are recommended for use with asterisk? |
| 22:39.30 | hmmhesays | #/lib/modules/2.6.18-53.1.4.el5xen/misc/ztdummy.ko |
| 22:39.58 | kand | adeelin: the 7960 or 7940 are common |
| 22:40.52 | kand | adeelin: I would use Polycom if at all possible |
| 22:41.04 | adeeln | kand: i've also noticed that there 7941/7961/7970/7971...would they all work? |
| 22:41.13 | lazybrain | how do you turn sip debug off ? |
| 22:41.21 | [TK]D-Fender | adeeln, Cisco is considerably lower down the list.... |
| 22:41.22 | outtolunc | sip no debug |
| 22:41.33 | lazybrain | thanks im a bit tired |
| 22:41.41 | [TK]D-Fender | lazybrain, "help" <- start reading |
| 22:41.51 | hmmhesays | hmm any ideas? |
| 22:42.12 | [TK]D-Fender | hmmhesays, "uname -a" |
| 22:42.40 | hmmhesays | you want my kernel version? |
| 22:42.49 | adeeln | [TK]D-Fender: yeah i know, but i have a client who doesn't want anything but cisco phones |
| 22:42.58 | hmmhesays | it built and is there |
| 22:43.08 | lazybrain | Tk d fender, you flood my console with sip debug then you tell me rtfm. Can you ease up a bit. Were adults here |
| 22:43.16 | [TK]D-Fender | adeeln, "that's nice". Their loss in more ways than one |
| 22:43.49 | adeeln | [TK]D-Fender: i'm aware |
| 22:43.50 | hmmhesays | #/lib/modules/2.6.18-53.1.4.el5xen/misc/ztdummy.ko uname -a 2.6.18-8.el5xen hmm good call |
| 22:43.54 | hmmhesays | but why would it even build |
| 22:44.07 | [TK]D-Fender | lazybrain, I handed you the manual rather than telling you to look for it. A somewhat important difference. it would have spat the answer clear as day and all you'd have had to do is look. |
| 22:44.16 | adeeln | hmmhesays: i've run into problems with that, on a different setup...i could never get it resolved either |
| 22:44.28 | [TK]D-Fender | hmmhesays, mismatched headers |
| 22:44.53 | hmmhesays | you'd think it would error out when you built it |
| 22:45.19 | [TK]D-Fender | hmmhesays, only when you try to load it IIRC |
| 22:48.06 | hmmhesays | bah why would yum grab the wrong hearders |
| 22:48.26 | hmmhesays | thats not annoying |
| 22:48.36 | [TK]D-Fender | hmmhesays, something symlinked to them? |
| 22:49.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:49.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:49.12 | hmmhesays | it seems as though the development package for this kernel doesn't exit |
| 22:49.14 | hmmhesays | *exist |
| 22:53.11 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 22:53.20 | mvanbaak | maybe you should update the kernel then ? |
| 22:53.51 | [TK]D-Fender | hmmhesays, or maybe just make sure they match.... |
| 22:54.30 | lazybrain | kand - I got *XXX working for most extensions but, when I press *10 the phone says reorder wierd |
| 22:54.42 | NovceGuru | hey, 1500/1500 192.168.66.132 D 20798 OK (474 ms), is that right on a 1 lan hop? :( |
| 22:54.45 | lazybrain | I dont see anything conflicting with *10 |
| 22:55.53 | kand | lazybrain: the _*XXX pattern is ony for anything that is * and three digits so it will not match *10 you could us _*XX. and read up on asterisk patterns |
| 22:56.21 | NovceGuru | something also bothering me is I have * running behind a NAT without any portfowarding and it can recieve calls from the upstream sip peer :| |
| 22:56.27 | lazybrain | kand - I was trying to dial *101 for example |
| 22:57.23 | kand | lazybrain: pastebin sip debug |
| 22:57.33 | *** join/#asterisk ExplodingLemur (n=change@204.16.141.196) |
| 22:59.06 | ExplodingLemur | Anyone interested in helping with a NAT issue? |
| 22:59.06 | ExplodingLemur | Several Snom 320 phones behind a NAT box, * machine on a public IP elsewhere. All the phones have the ability to dial out, but only sporadically can any receive calls from the * box. I've got all the phones checking an STUN server, and nat is set to yes on the * box for all the phones. |
| 22:59.53 | kand | ExplodingLemur: try changing your signaling to different ports (5060,5061,5062,ect) |
| 23:00.56 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 23:00.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:01.00 | [hC] | this is interesting |
| 23:01.08 | ExplodingLemur | kand: as the source port for each phone, or as the destination port for each phone to hit on the * box? |
| 23:01.10 | [hC] | specifying accountcode in iax.conf does not apply it to the cdr when a call goes TO a peer? |
| 23:01.15 | [hC] | only when a call comes FROM a peer? |
| 23:03.20 | kand | ExplodingLemur: I would do both. |
| 23:03.44 | *** join/#asterisk angom_h (n=Angel@201.170.62.90) |
| 23:03.46 | kand | ExplodingLemur: The idea is to make the NAT unique for each phones ip |
| 23:04.35 | lazybrain | kand - there is no much info to paste the sip debug to pastebin |
| 23:04.59 | ExplodingLemur | kand: How would I get Asterisk to bind to multiple ports to listen for SIP? |
| 23:06.15 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:06.37 | kand | lazybrain: Paste what you got ExplodingLemur: one sec |
| 23:07.03 | lazybrain | I believe its ports =1,2,3 etc |
| 23:07.06 | *** join/#asterisk russellb_ (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:07.06 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:08.13 | *** topic/#asterisk by russellb_ -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.15 (2007/11/29), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.7.1 (2007/12/13), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 23:11.24 | lazybrain | kand - could this be why ? exten => _1[0-9][0-9],1,Macro(extension-internal,${EXTEN}); |
| 23:12.09 | kand | ExplodingLemur: sorry, I thought we had bound the ports in asterisk but we used iptables to translate (public side) |
| 23:13.19 | ExplodingLemur | gotcha...yeah, found that in a mailing list entry just now |
| 23:13.28 | ExplodingLemur | I did discover keepalive settings on the phones, I've set those and will test some... |
| 23:13.35 | kand | lazybrain: That is messy but should link to the macro. I cant help if your just going to give me pieces. I think you need to go back and read..... |
| 23:15.26 | kand | ExplodingLemur: It has been my experience that those help but dont correct the issue you are describing. The sure fire way I recommend is to replace the router, I like to use http://www.amazon.com/Netopia-Broadband-Router-4-Port-Switch/dp/B0009VU1JK/ref=pd_bbs_8?ie=UTF8&s=electronics&qid=1197587706&sr=8-8 |
| 23:16.14 | kand | It has diffserv too which helps makes a huge improvement. |
| 23:16.27 | hmmhesays | ugh finally |
| 23:16.30 | hmmhesays | I hate it when other people do the linux installs |
| 23:17.22 | ExplodingLemur | kand: superior NAT implementation? (currently they have a little Linksys box) |
| 23:19.03 | *** join/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 23:19.05 | kand | ExplodingLemur: first thing I do at every new client is recommend this baby, ya MUCH better NAT, enterprise class features, SNMP (which I link to Zenoss), diffserv... you name it |
| 23:19.28 | kand | ExplodingLemur: If they wont shell out then I go with the port trick |
| 23:20.18 | hmmhesays | which? |
| 23:20.39 | kand | which nat? |
| 23:21.09 | hmmhesays | no recommend what? |
| 23:21.47 | kand | Netopia routers whose model ends in ENT (enterprise). Only a few more bucks then SOHO and well worth it |
| 23:21.58 | *** join/#asterisk asr33 (n=island@ppp-RAS1-5-233.dialup.eol.ca) |
| 23:21.58 | hmmhesays | gotcha |
| 23:22.35 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:22.54 | *** join/#asterisk admgecko (n=admgecko@adsl-87-102-68-32.karoo.KCOM.COM) |
| 23:22.57 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 23:24.28 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 23:27.05 | hmmhesays | finally got this fscking install done, what a nightmare |
| 23:27.19 | ExplodingLemur | kand: I'll have more control over their network once they move to a new office, we'll have Cisco devices on either side with connectivity from two different providers to our datacenter...that should work much more nicely. |
| 23:28.06 | admgecko | is anyone a wireless access point with a phone, to make a normal phone, wireless? |
| 23:28.12 | admgecko | *using |
| 23:29.59 | kand | ExplodingLemur: that should do it... |
| 23:30.26 | ExplodingLemur | hopefully the keepalive trick will work until the move. Thanks for the help! |
| 23:30.59 | kand | np |
| 23:34.49 | *** part/#asterisk asr33 (n=island@ppp-RAS1-5-233.dialup.eol.ca) |
| 23:35.18 | [hC] | does asterisk not apply accountcode to cdr when calling TO an IAX peer? I seem to only have accountcodes in the database when a call comes to me FROM the iax peer? |
| 23:38.08 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 23:39.26 | mltlnx | [TK]D-Fender: The other day you gave me a clue on how to disable MOH for a certain context..What was that clue again? |
| 23:39.48 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 23:40.47 | *** join/#asterisk MichaelJE2 (i=MichaelJ@mikey.hacks.xcelor8.net) |
| 23:46.43 | [hC] | nobody knows this accountcode behavior? |
| 23:47.28 | outtolunc | do you have an accountcode set for whatever tech type you are using to dial out |
| 23:47.41 | outtolunc | (probably not) |
| 23:47.52 | [hC] | in the iax peer, lets call it 'supercustomer' I have accountcode=supercustomer |
| 23:48.03 | [hC] | when a call comes from them to me, (and subsequently out my pri) i get the account code |
| 23:48.10 | [hC] | when a call comes in my pri, and goes via iax TO them, no accountcode. |
| 23:48.44 | outtolunc | the accountcode you get on a PRI is set in zapata.conf |
| 23:49.04 | [hC] | er |
| 23:49.16 | [hC] | so this is what im asking |
| 23:49.23 | [hC] | accountcode is always trumped by the incoming technology? |
| 23:49.39 | [hC] | it is not set when i call TO supercustomer, based on their accountcode in iax.conf? |
| 23:50.17 | outtolunc | a call down a pri has no idea what 'accountcode' some channel type XYZ did 4 boxes away |
| 23:50.31 | outtolunc | short answer, no |
| 23:50.37 | [hC] | its not 4 boxes away |
| 23:50.41 | outtolunc | haha |
| 23:50.51 | [hC] | we're talking about one asterisk box here |
| 23:50.58 | [hC] | pri -> * -> IAX to someone else |
| 23:51.05 | [hC] | the * box just does handoffs |
| 23:51.26 | [hC] | so what this tells me then, as far as cdr's go, is the account code is always set by the originator |
| 23:51.47 | [hC] | or rather, asterisk pulls the account code from the configuration file relative to the originating source |
| 23:52.03 | [hC] | ie if a call comes in PRI, it uses acccountcode for PRI, if it comes in IAX, accountcode from iax peer, etc. |
| 23:52.24 | [hC] | the point being it never applies it from even the peer you dial TO... in a handoff event |
| 23:52.31 | [hC] | so i would have to put this in using the dial plan.. |
| 23:52.36 | outtolunc | the point was that a PRI can't 'pass' the accountcode, it can only ADD it from what it knows (in zapata.conf) |
| 23:53.04 | [hC] | im not talking about 'passing' accountcode |
| 23:53.33 | [hC] | nor the pri adding the accountcode.. asterisk clearly has to add it based on a set of rules |
| 23:53.46 | [hC] | what i needed to know was that asterisk uses the accountcode setting only from the originating technology's config |
| 23:53.52 | [hC] | (zapata.conf, iax.conf, sip.conf, etc) |
| 23:53.54 | outtolunc | i feel like i'm walking in circles |
| 23:54.08 | [hC] | you've already answered my question, heh. |
| 23:55.14 | [hC] | Im just saying that i didnt understand that by accepting a call on a pri, and having that call dial some one via IAX, and having that IAX peer have an accountcode set, i expected that the account code of the iax peer that i sent the call to would be put in the CDR, but it is now clear that asterisk only uses the accountcode from the incoming path |
| 23:55.22 | [hC] | that should probably be made clear in the sample configs, or something |
| 23:55.32 | [hC] | to clarify how accountcode is used.. |
| 23:58.03 | DoDaT69 | is there any open source software out there for a call center scenario |
| 23:58.31 | DoDaT69 | more specifically the attendant, where a did rings and we can pop up customizable scripts on the screen? |
| 23:58.33 | [hC] | Yes, vicidial |
| 23:59.02 | DoDaT69 | friggin awesome!!!!! |
| 23:59.05 | DoDaT69 | thanks so much man! |
| 23:59.21 | DoDaT69 | have you used it ? |