00:02.41 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
00:05.41 | *** join/#asterisk kimosabe (n=kimosabe@dsl-201-133-195-203.prod-infinitum.com.mx) |
00:05.48 | *** join/#asterisk SPoon_TSX (n=klee@h24-83-96-211.sbm.shawcable.net) |
00:06.04 | SPoon_TSX | Hello there, For all of the experts here. I got a quick question. |
00:06.12 | eric_s | night |
00:06.16 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
00:06.42 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
00:06.43 | SPoon_TSX | I have an asterisk installed and everything works okay except I got some whitenoise at the background when I make a PSTN call. Sip to SIP is prefect. |
00:06.57 | SPoon_TSX | I am wondering what could possible causing the problem? |
00:07.49 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
00:07.53 | SPoon_TSX | anyone? |
00:08.46 | kimosabe | i have a tdm400p card and it gives dial tone and i have a sip account on a remote server any help me with a config example so that i can dial via tdm400p using sip acount |
00:09.29 | kimosabe | is any one here from selectfone ?? |
00:14.27 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
00:14.59 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
00:15.14 | harryvv | What ntp servers do you all use for your phones? |
00:18.48 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
00:19.20 | [av]bani | my own |
00:19.32 | [av]bani | do not abuse happy fun public ntp servers |
00:19.49 | Saturn-- | I have musiconhold setup |
00:19.56 | Saturn-- | i can play an mp3 with "mp3player()" |
00:20.04 | Saturn-- | However when i try and put a call on hold it just says |
00:20.09 | Saturn-- | started music on hold |
00:20.11 | Saturn-- | then right away |
00:20.14 | Saturn-- | stopped music on hold |
00:20.16 | Saturn-- | no explanation |
00:20.43 | Saturn-- | lewloal |
00:21.52 | harryvv | naa, just looking for one other then redhat. |
00:22.23 | Saturn-- | Trying to figure out why it would do this |
00:22.49 | brodiem | harryvv, ntpd. |
00:23.04 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
00:25.18 | generalhan | anyone in here using multiple digium cards in the same server ? like a TE card and a TDM card ? |
00:25.34 | harryvv | brod I know its technically called ntpd but thats not how is talked about in general conversation or even a google search. |
00:26.06 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
00:31.27 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
00:33.36 | harryvv | btw, anyone ever get cwcn running on a ip500? |
00:34.15 | DoktorGreg | can anyone post a link to get me started on sorting out music on hold and ivr crackling problems? |
00:34.50 | harryvv | sounds like a ground issue. mabey bad mic problem when recording it? |
00:35.12 | DoktorGreg | default sounds, full drop outs... |
00:35.23 | DoktorGreg | both soft and hard phones are all working perfectly to each other... |
00:35.53 | DoktorGreg | only getting the static when i hit voicemail, ivr and moh |
00:36.10 | DoktorGreg | the emailed voice mails seem to sound ok |
00:36.35 | harryvv | odd |
00:36.52 | hinckc | any relation to server load? does it happen on when only 1 user is there? |
00:37.13 | DoktorGreg | yup... any relation to sound card drivers??? |
00:37.15 | Darwin35 | ~amp |
00:37.17 | jbot | [amp] "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
00:37.48 | k-man | anyone know if there is a tapi driver for windows that can direct a call through voip? |
00:38.15 | tzanger | man I love when people who don't have clue try to sound technically ept in -users |
00:38.33 | tzanger | that whole te110 interrupt problem thread |
00:38.46 | *** part/#asterisk The_Blob (n=jcragg@24.85.224.102) |
00:39.00 | tzanger | the guy's got a local apic and the cards are on differnet APIC IRQs but the one guy says to turn off APIC to get them on "real" irqs... heh |
00:39.29 | tzanger | dude's problem is that he has three digium cards in one box, the interrupt load is pretty high (3000 interrupts/sec not counting anything else) |
00:40.21 | generalhan | tzanger: i have 2 cards in one system and im having some issues ... is this common ? |
00:40.43 | DoktorGreg | digium cards are really really reluctant to work properly and share irq's |
00:40.51 | tzanger | generalhan: high interrupt load isn't cool |
00:40.57 | generalhan | well i checked on that .. they arent charing the IRQs |
00:41.01 | tzanger | DoktorGreg: I have next to no issues doing so |
00:41.21 | generalhan | well i really dont have a choice i have a TE210 for the PRI lines ... and i have a TDM for my fax machines |
00:41.24 | tzanger | on moderate hardware, yes, of course. WIth shitty products you're sharing IRQs with, of course... |
00:41.36 | tzanger | but if you have good drivers sharing interrupts ins't too much of an issue with digium cards |
00:41.57 | generalhan | well * refuses to load my TDM card stuff ... it just wont do it |
00:41.59 | DoktorGreg | what mobo do you suggest tzanger? |
00:42.23 | generalhan | and im working on a production server so i have to wait until after 8pm MST to shut down the lines to test some things |
00:42.40 | harryvv | anyone here on the west coast using a ntpd server thay find reliable and open? |
00:42.48 | tzanger | generalhan: it shoudl load, but work like shit |
00:43.22 | generalhan | it wont do it i dont know why ... but im testing some different things out |
00:43.30 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe205.t-net.net.ve) |
00:43.35 | DoktorGreg | what about digium on sunfire servers? |
00:43.48 | generalhan | like for some reason when i load the wct4xxp it gives me an error about my TDM card ... so im not sure what that is all about |
00:44.10 | generalhan | i checked zapata.conf and zaptel.conf like 1000 times to see what was happening and it looks good to me |
00:44.31 | harryvv | generalhan how many phones in your network? |
00:44.39 | Saturn-- | any suggestions on why the musiconhold would "stop" like immedately after it starts, and never play |
00:44.39 | Saturn-- | ? |
00:45.05 | harryvv | Saturn-- do a top and play the vm and see what happens. |
00:45.07 | generalhan | 30-40 Aastra SIP phones and 15 Cisco 7960s |
00:45.15 | harryvv | also turn on cli in asterisk |
00:45.20 | tzanger | generalhan: ohh |
00:45.22 | generalhan | why do you ask |
00:45.23 | harryvv | that is, look at cli |
00:45.32 | Saturn-- | what |
00:45.32 | harryvv | generalhan just curios |
00:45.35 | tzanger | it's complaining because the default config runs ztcfg after the module load |
00:45.40 | Saturn-- | i am using the cli |
00:45.46 | tzanger | and you didn't load BOTH drivers so ztcfg will fail out because some channels are missing |
00:45.54 | Saturn-- | voicemail isn't related, i donte ven have it on |
00:45.58 | harryvv | does it return errors in cli when playing vm? |
00:46.07 | harryvv | whats the issue then |
00:46.14 | Saturn-- | not voicemail, musiconhold |
00:46.21 | generalhan | well i made a script so that i wouldnt get messed up ... it loads zaptel, then wct4xxp then wctdm |
00:46.28 | Saturn-- | it just says "Started music on hold...." |
00:46.35 | Saturn-- | then "Stopped music on hold" |
00:46.38 | Saturn-- | immediately after |
00:46.40 | Saturn-- | no error, nothing |
00:47.05 | tzanger | generalhan: yes, but if your modules.conf still says "post-install /path/to/ztcfg" it'll error out |
00:47.48 | generalhan | hmmm |
00:48.52 | generalhan | in modules.conf i just have autoload=yes |
00:49.08 | harryvv | Saturn-- first install? |
00:49.10 | generalhan | and load=>res_musiconhold.so |
00:49.17 | tzanger | no /etc/modules.conf or /etc/modprobe.conf (the kernel module config stuff, not asterisk) |
00:49.27 | generalhan | ohh |
00:49.58 | generalhan | install wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg |
00:50.03 | generalhan | that stuff ? |
00:50.59 | Qwell | Saturn--: You need to install mpg123. |
00:51.19 | Qwell | Saturn--: from the asterisk source dir, type `make mpg123; make install` |
00:51.54 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.94.telnor.net) |
00:52.15 | tzanger | generalhan: yeah, or just erase the crap in modules.conf and do it manually :-) |
00:52.19 | tzanger | like a real man :-) |
00:52.29 | generalhan | haha "real man" he says ! |
00:52.49 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
00:53.45 | generalhan | i dont know enough about linux to fill a cup ... you say manual ANYTHING to me and my answer is no ! hahaha ! |
00:53.45 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
00:53.56 | tzanger | :-) |
00:54.19 | generalhan | ok well i cant do anything while my reps are still on the phone ... so im going home to wait till after hours to test some stuff |
00:54.24 | generalhan | ill be back then guys !! |
00:54.28 | generalhan | thanks for all the help ! |
00:55.11 | Saturn-- | Qwell |
00:55.17 | Saturn-- | i just installed mpg123 |
00:55.22 | Saturn-- | the mp3player() function works |
00:55.24 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
00:55.29 | justinu | qwell |
00:55.31 | justinu | fix my shit |
00:55.39 | Qwell | justinu: fix mine |
00:55.40 | generalhan | justinu: HAHAHAHA |
00:55.46 | Saturn-- | but the musiconhold still does not |
00:55.47 | Qwell | or pay me, newb |
00:55.48 | generalhan | Qwell: HAHAHAHAHAHAHAHAHAA |
00:55.48 | Qwell | :P |
00:56.08 | Qwell | Saturn--: Did you install mpg123 the way I said? |
00:56.13 | Saturn-- | yes |
00:56.19 | Qwell | and you restarted *? |
00:56.22 | Saturn-- | i probably have the config wrong |
00:56.22 | *** join/#asterisk Eggplant (i=No@dsl-469.cascadeaccess.com) |
00:56.22 | Saturn-- | yes |
00:56.44 | *** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
00:59.03 | Saturn-- | But it doesn't indicate any trouble with the config |
01:05.51 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
01:06.15 | Jaxxan | you dont need mpg123 with 1.2.x versions right ? |
01:06.23 | Qwell | Jaxxan: "need", no |
01:06.25 | Qwell | but you need something |
01:06.40 | Jaxxan | thought asterisk 1.2.x had it's own player |
01:06.49 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
01:06.54 | Qwell | there is format_mp3 in asterisk-addons |
01:07.00 | Jaxxan | dont get me wrong, i think i'm still using mpg123 |
01:07.39 | justinu | good... i was about to get you wrong |
01:07.54 | harryvv | anyone here have a ip500 and know what it takes for a Caller Waiting party to ring the other line ? |
01:08.15 | Jaxxan | i upgraded from 1.0.9 to 1.2.6 though, that's why i have mpg123 setup, i was just reading something about 1.2.x having it's own player is all. just wondering. |
01:08.44 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
01:09.07 | Jaxxan | i have a problem with my queue_log file. it's not recording callerid(name) |
01:09.19 | Saturn-- | it can play with mpg123 i am assuming, for mp3player function, just onhold it feels it doesn't want to for some reason |
01:09.19 | Saturn-- | ohwell |
01:09.23 | Jaxxan | it only shows callerid(number) |
01:09.44 | Jaxxan | and i need it to show the calleridname |
01:09.58 | Jaxxan | well, i need it to print it in the queue_log |
01:10.14 | Jaxxan | it printed when i was using 1.0.9, but 1.2.6 doesn't print it. |
01:10.44 | Jaxxan | and it sucks cause with the way i do reporting with queuemetrics now |
01:11.17 | Jaxxan | i label calls based on callerid rather than separate queues and now i can't see what kind of call it was |
01:11.43 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
01:11.56 | Jaxxan | unless i parse the cdr, but that doesn't do me any good. |
01:12.23 | justinu | Saturn--: did you make sure you answered the call? |
01:12.34 | Saturn-- | yeah |
01:12.42 | Jaxxan | how do i go about getting that changed ? |
01:12.44 | Saturn-- | if i didn't it would just keep ringing |
01:12.54 | Saturn-- | even though CLI shows the proc as running |
01:12.54 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
01:14.50 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
01:20.39 | justinu | Saturn--: got ztdummy loaded? |
01:21.00 | Saturn-- | isn't working on NetBSD |
01:21.11 | Saturn-- | I got it to play music, but only if there's like incoming noise |
01:21.18 | Saturn-- | if I am muted or silent, the music stops |
01:21.20 | Saturn-- | until i make more noise |
01:21.21 | Saturn-- | i dont get it |
01:21.29 | Qwell | Saturn--: turn off VAD/Silence Suppression on your phone |
01:21.30 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
01:22.00 | Qwell | ~vad |
01:22.01 | jbot | [vad] Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
01:22.09 | Saturn-- | thanks, perfect |
01:22.21 | Saturn-- | except for being choppy, but what can I expect from california to new york |
01:22.24 | Saturn-- | thanks |
01:22.34 | Qwell | You can expect...perfection |
01:23.29 | Nugget | * Void where prohibited by law |
01:24.51 | Saturn-- | I better read more then |
01:24.53 | Saturn-- | to tweak this up |
01:25.11 | *** part/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
01:25.32 | Saturn-- | also the encoding of the mp3's isn't helping |
01:27.07 | Saturn-- | but that i know how to fix |
01:27.50 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
01:28.08 | cced | Jaxxan :hi |
01:28.24 | Jaxxan | yo |
01:29.02 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
01:29.51 | Jaxxan | how goes the documentation hunt |
01:30.18 | cced | doing now .Jaxxan. |
01:30.29 | file | Nugget: okay that was baddddddddddd |
01:30.34 | Nugget | moo? |
01:30.50 | file | Nugget: mooooooooooo |
01:31.00 | Nugget | apnp 'oow umop apisdn ue w,i |
01:31.02 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
01:31.18 | cced | Jaxxan: I want to draw pri state machine figure. |
01:31.23 | Qwell | Nugget: That was painful |
01:31.34 | cced | which figure I can use in Q.921 931 spec? |
01:32.46 | Jaxxan | no clue |
01:32.53 | Jaxxan | that's to technical for me |
01:33.00 | Jaxxan | i know know how to get it to work |
01:33.05 | Jaxxan | erm, i just know |
01:33.50 | Jaxxan | my boss said, make it work and i did. |
01:34.02 | justinu | why do you want to draw a pri state machine figure? |
01:34.08 | file | so Qwell |
01:34.43 | cced | faint. yes .it work well.~~ |
01:35.27 | Lino` | lol |
01:35.35 | Lino` | verizon might want to buy vodafone.airtouch |
01:36.10 | Lino` | whole vodafone |
01:36.19 | Qwell | so file |
01:36.30 | file | no soup for you |
01:36.36 | *** mode/#asterisk [+o file[laptop]] by file |
01:36.43 | Qwell | ack! |
01:36.56 | file | I won't hurt you! |
01:36.59 | Qwell | Would have been FAR better, as a kickban message. |
01:37.00 | Qwell | sheesh |
01:37.17 | file | bat! |
01:37.31 | file | Qwell: translation? |
01:37.38 | file | Build, Asterisk, Test! |
01:38.04 | Qwell | why test? |
01:38.56 | file | why not?!? |
01:39.12 | Qwell | because it compiles |
01:39.34 | Corydon76-home | It compiles! Ship it! |
01:39.43 | Qwell | exactly |
01:39.50 | file | crazy crazy people |
01:40.10 | Corydon76-home | Pot. Kettle. Black. |
01:40.16 | Qwell | I use the following development model: |
01:40.30 | file | oh. |
01:40.33 | Qwell | code, compile, test, code, compile, test, code, compile, release |
01:41.12 | Corydon76-home | Qwell version 1.0.0.368b19 |
01:43.32 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
01:43.46 | Qwell | at work, I probably compile every...10 minutes, tops |
01:44.27 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
01:46.58 | *** join/#asterisk Myconid3 (n=myconid@69-164-122-221.sbtnvt.adelphia.net) |
01:46.59 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
01:47.42 | techman97_andy | hey all again...new time of day, new people, hopefully some new ideas. I'm a 100% SIP environment (SIP xLite for clients, SIP provider for PSTN access), and I'm having an issue where I can dial (or receive) the call, but the RTP traffic won't flow until I place the caller on hold and pull them back off (as quickly as a double click). I've tested this with the same results from different networks / extensions, and it's alway |
01:48.18 | Myconid3 | what is the most mature commercial asterisk product |
01:48.36 | tainted- | Myconid3 for doing what |
01:48.53 | Qwell | tainted-: asterisking |
01:48.55 | Myconid3 | Running in the enterprise |
01:49.00 | justinu | techman97_andy: capture the network traffic and analyze it with ethereal. figure out why rtp doesn't flow by looking at the SIP and SDP messages |
01:49.04 | Myconid3 | 6 sites, 170 clients |
01:49.18 | Myconid3 | "no dropped calls" is concern #1.. |
01:49.23 | Myconid3 | which im not sure asterisk can provide |
01:49.25 | techman97_andy | justinu: Can you give me some guidance on how to set that up (never used ethereal) |
01:49.29 | tainted- | i would have to say Qwell is a pretty mature product of asterisk |
01:49.40 | Qwell | ~qwell |
01:49.44 | jbot | you are, like, a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
01:49.44 | tainted- | lol |
01:50.03 | Qwell | synergistics, baby |
01:50.11 | tainted- | techman97_andy try with other softphones |
01:50.14 | justinu | techman97_andy: use "tcpdump -s0 -w filename.cap" to capture the traffic on your * server |
01:50.29 | justinu | transfer the cap file to your workstation and load it into ethereal |
01:50.39 | techman97_andy | where can I get ethereal? is it freeware? |
01:50.40 | tainted- | Myconid3 sounds like the project is over your head frankly.. outsource it |
01:50.56 | justinu | google |
01:50.57 | justinu | it's free |
01:50.59 | techman97_andy | k |
01:51.02 | techman97_andy | brb |
01:51.02 | Myconid3 | thanks for your expert analysis. |
01:51.12 | Qwell | Myconid3: Do you have $1 million to spend? |
01:51.20 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
01:51.22 | Myconid3 | Qwell: if we did, we wouldnt be running Asterisk ;) |
01:51.23 | Qwell | Because if not, you're going to have dropped calls, with any solution |
01:51.29 | [hC] | Myconid3: good idea, ask for help then be a cock. |
01:51.31 | cced2 | <cced> Qwell : which part are yo familiar?libpri zaptel asterisk? |
01:51.41 | Myconid3 | Qwell: We dont have any with our current nortel pbx. |
01:51.41 | tainted- | no dropped calls have never been a concern for anyone involved with asterisk or VoIP for that matter |
01:51.42 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
01:51.43 | Qwell | cced: asterisk |
01:51.52 | Qwell | [hC]: You! |
01:51.58 | Qwell | [hC]: Where are my test results? :P |
01:51.59 | [hC] | Qwelzz!@#13123111oneone |
01:52.00 | cced2 | yes. chan_zap chan_sip channel.c |
01:52.09 | [hC] | Sorry, I had a friend over for the weekend |
01:52.12 | [hC] | and nerdery ceased progress. |
01:52.18 | Qwell | excuses, excuses |
01:52.21 | [hC] | :) |
01:52.29 | [hC] | When I go home tonight, I'll load up your channel driver |
01:52.35 | [hC] | now that my phone is running sccp |
01:52.44 | Qwell | cool |
01:52.48 | Myconid3 | so asterisk is still in its infancy? |
01:52.49 | cced2 | Qwell: I want to drwa SIP state diagrams |
01:53.03 | [hC] | I hate the SCCP firmware for 7960 compared to the SIP firmware |
01:53.16 | tainted- | Myconid3 not if u know what you're doing |
01:53.26 | Qwell | [hC]: because you don't have a good skinny channel driver to use :p |
01:53.27 | [hC] | Few reasons: Blind Transfers. Call Forwarding (maybe just chan_sccp's way) |
01:53.30 | Qwell | sccp rocks |
01:53.35 | [hC] | well |
01:53.37 | Qwell | yes, chan_sccp cfwd is stupid |
01:53.43 | [hC] | on chan_sccp, i already want to switch back its that bad |
01:53.47 | Qwell | can only use it offhook...wtf is that? |
01:53.48 | [hC] | however on the 7970 its awesome |
01:53.51 | [hC] | yeah |
01:54.01 | Qwell | mine will let you do it onhook :) |
01:54.06 | [hC] | and the blind transfer thing, apparently its a SIP hack to make it work like the way id expect on SIP |
01:54.12 | Qwell | huh? |
01:54.17 | [hC] | You cant blind xfer |
01:54.24 | Qwell | lame |
01:54.27 | [hC] | you either hit transfer the second time REALLY FAST or decide to speak to the person |
01:54.30 | cced2 | Qwell . how chan_zap read or write data from zaptel? |
01:54.39 | Qwell | [hC]: okay, I'll keep that in mind |
01:54.55 | Qwell | will try to think of a solution |
01:54.58 | [hC] | I asked sergio and he said that the only reason the sip firmware can do the 100% unattended transfer is because you can use sip headers to do it seamlessly |
01:55.08 | Qwell | well...he's an idiot. :) |
01:55.13 | [hC] | Also, Im not sure if this is the phones fault or what |
01:55.21 | [hC] | but the 7970 will NOT retain placed/received calls lists |
01:55.33 | Qwell | That is a firmware thing, I'm pretty sure |
01:55.38 | [hC] | <PROTECTED> |
01:55.46 | Qwell | weird |
01:55.54 | [hC] | I thought it may have been a locale thing since i get an error updating locale msg |
01:56.02 | [hC] | but I dont know how to correct that, Ive searched for the files with no luck |
01:56.10 | Qwell | Have you tried the 8.0 firmware? |
01:56.16 | [hC] | Yah |
01:56.22 | Qwell | same thing? |
01:56.27 | [hC] | Yepp |
01:56.38 | Qwell | strange |
01:56.43 | [hC] | Im pretty sure it asks for a td-sccp.jar at startup |
01:56.44 | [hC] | that I dont have |
01:56.47 | Gamercjm | Hmm can i have asterisk record my voice? like call my DID and have it record a msg for me |
01:56.48 | [hC] | and it may be related to that |
01:56.55 | [hC] | It seems to come from callmanager |
01:56.58 | [hC] | cause it didnt come with the firmware. |
01:57.18 | *** join/#asterisk Tili (i=Tili@61.140.191.181) |
01:57.22 | tainted- | i need some asterisk work done. i heard from a friend that it's really cheap and fast and should cost around $50.00 (including hardware). I need to connect 3 remote offices, ~80 lines and have 100% uptime. |
01:57.37 | Qwell | tainted-: heh |
01:57.53 | Ariel_ | $ 50.00 hummm |
01:58.06 | [hC] | tainted-: my friend said the same thing, and said that all you guys will help me do it for free, and you're all SUPER HELPFUL |
01:58.06 | [hC] | :) |
01:58.09 | Ariel_ | wow, phones along cost more then that. |
01:58.14 | Gamercjm | ill install asterisk for $50 ;) |
01:58.16 | Dream_WEaver | heh |
01:58.17 | Ariel_ | The software is free |
01:58.24 | Gamercjm | lol |
01:58.29 | Dream_WEaver | Install, yes. Configure - forget it :) |
01:58.47 | tainted- | well i heard there are like free phones (like skype) so don't try to rip me off |
01:58.56 | DoktorGreg | Ill install a complete asterisk system with 10 phones for 8k if anyone wants it |
01:58.56 | Gamercjm | well softphones |
01:58.58 | Gamercjm | are free |
01:59.02 | Ariel_ | tainted-, there are free softphones yes |
01:59.15 | Ariel_ | xlite works well |
01:59.20 | Ariel_ | and yes there free |
01:59.22 | DoktorGreg | xlite seems to work great |
01:59.29 | Myconid3 | tainted-: I have 70 7960's if that makes it any easier :P |
01:59.29 | Ariel_ | but you need to start doing some reading. |
01:59.31 | Ariel_ | ~docs |
01:59.33 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
01:59.33 | DoktorGreg | iaxComm works well also, but is fugly |
01:59.36 | Dream_WEaver | tainted: Clearly you have been mislead if you thought there were some people here that will install your system cheap and effectively :) |
01:59.45 | Dream_WEaver | This is more of a support channel than anything. |
02:00.17 | DoktorGreg | how do you deploy software to those 7960's? |
02:00.26 | Jaxxan | Anyone have any idea how to get the callerID Name to write to the queue_log ? |
02:00.28 | tainted- | omg!!11 so ungreateaful 4 work!!!!!!11 |
02:00.29 | Ariel_ | 7960 software tftp |
02:00.33 | Myconid3 | DoktorGreg: cisco call center :~) |
02:00.36 | DoktorGreg | and if you dont mind me asking... |
02:00.43 | DoktorGreg | what kind of software do you run on em? |
02:00.50 | Qwell | DoktorGreg: the firmware |
02:00.55 | [hC] | cisco's firmware. |
02:00.56 | Qwell | That's all you run "on" them |
02:01.06 | Dream_WEaver | Cisco - the one that doesn't account for DST |
02:01.17 | Dream_WEaver | Unless you pay extra for the firmware upgrade. |
02:01.21 | Ariel_ | polycom, polycom's |
02:01.25 | Dream_WEaver | (or it comes with the phone) |
02:01.28 | Dream_WEaver | like polycom |
02:01.47 | DoktorGreg | so its a phone with a big color screen and ldap support??? |
02:02.06 | Myconid3 | docelm0[QUOTE=BlackMaxima21]my friend has a 2002 se automatic and has this weird shifting problem from 2nd to 3rd. it doesn't want to shift right away, if you keep your foot on the gas pedal the rpms keep going up but your not accelerating and then they drop down when it shifts into 3rd. Do you guys have an idea of what this could be??[/QUOTE] |
02:02.08 | Myconid3 | erm. |
02:02.11 | Myconid3 | stupid pastey. |
02:02.38 | cced2 | Hi Ariel_ |
02:36.26 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
02:36.26 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support |
02:36.35 | Myconid3 | omg.. i just downloaded the stupid mac torrent of command and conquer.. grawr. |
02:37.26 | techman97_andy | hmmmm |
02:37.49 | DoktorGreg | how do i tell what is a good setting for my jitter buffer/. |
02:37.52 | DoktorGreg | ? |
02:38.05 | techman97_andy | I have iptables on the asterisk server, allowing UDP 10000-20000 through. I see that the RTP packets from my provider are 20574...does that make sense? |
02:38.18 | justinu | src port 20574? |
02:38.43 | techman97_andy | Got RTP packet from 64.xx.xx.xx:20574 (type 0, seq 65279, ts 200960, len 160) |
02:38.43 | techman97_andy | Sent RTP packet to 70.xx.xx.xx:10000 (type 3, seq 9500, ts 13280, len 33) |
02:38.57 | techman97_andy | 70 is me, 64 is my SIP provider |
02:39.00 | justinu | you iptables rule should be allowing packets to dest port 10000-20000 |
02:39.02 | techman97_andy | that was from the CLI |
02:39.11 | techman97_andy | yeah, I'm doing that in iptables |
02:39.35 | justinu | the question is, do you get any RTP from the provider before the phone goes on hold? |
02:39.46 | justinu | or from the phone |
02:39.48 | techman97_andy | nope. no RTP at all until the hold/offhold thing |
02:39.57 | justinu | vverified that w/ ethereal right? |
02:40.03 | techman97_andy | as far as I can tell, yes. |
02:40.30 | justinu | well... i dunno... asterisk won't send RTP unless it gets any |
02:40.44 | Jaxxan | if i'm using agents.conf to record agent calls as gsm, how can i increase the volume of the file? |
02:40.49 | Jaxxan | it's really low |
02:40.55 | Jaxxan | or should i be using a different format ? |
02:40.57 | techman97_andy | so maybe that 20574 udp port that my provider is sending is being regarded as unsolicited by iptables... |
02:40.58 | techman97_andy | ? |
02:41.06 | justinu | if your behind a nat, possibly |
02:41.19 | techman97_andy | my SIP client is behind a NAT, server is just ipchains. |
02:41.25 | techman97_andy | and ipchains is on the * server |
02:41.36 | techman97_andy | my SIP provider is public |
02:41.37 | justinu | server is public ip? |
02:41.44 | techman97_andy | server is public IP - long story. |
02:41.47 | justinu | k |
02:42.08 | justinu | well... you say you're not recieiving any rtp from the provider until the phone goes on/off hold? |
02:42.14 | techman97_andy | correct. |
02:43.00 | justinu | whats weird is the softphone should start sending rtp as soon as it goes off hook |
02:43.03 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
02:43.10 | justinu | that should be enough to kick off the whole chain |
02:43.22 | cced | do zaptel read or write data from hardware using DMA OR mmap? |
02:44.03 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
02:44.18 | techman97_andy | I may have misstated myself - let me back up. From my understanding of SIP phones, *, and SIP providers, is that I go offhook, dial the number...* places the call on my behalf and acts as an RTP "funnel" between the provider and my softphone, correct? |
02:44.55 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
02:44.57 | techman97_andy | but even taking that into account, SIP softphone to SIP softphone within Asterisk still has the same problem....hold/offhold, and we're good. |
02:45.24 | [av]bani | http://www.beckysweb.co.uk/beckysblog/2006/03/conversational-ebonics.asp |
02:45.30 | techman97_andy | my environment is the Asterisk server has a public IP, and I have softphones that connect from their homes / private offices....so everyone is flippin' NAT'd. |
02:45.48 | *** join/#asterisk file (n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net) |
02:46.38 | Hmmhesays | well guys wish me luck |
02:46.39 | Hmmhesays | i'm off |
02:46.48 | techman97_andy | bye bye |
02:47.08 | justinu | techman97_andy: rtp proxy is term |
02:47.31 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
02:47.32 | justinu | again, the problem is with the softphone |
02:47.48 | justinu | a real phone would start sending rtp right as it goes off hook |
02:48.05 | justinu | i've used xlite and not had that problem, so it must be a config issue |
02:48.14 | justinu | but xlite has a lot of options, so have fun... |
02:48.29 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
02:48.43 | techman97_andy | This has to be a blocked port thing...I initially set this whole thing up in one location on the same subnet and everything was cool |
02:48.58 | techman97_andy | as soon as I seperated and distributed, I've been having problems like this...:S |
02:49.17 | justinu | well, if the phones are on the same lan, doubtful |
02:49.35 | justinu | perhaps xlite is using stun and incorrectly determining it's IP |
02:49.53 | *** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net) |
02:53.54 | [TK]D-Fender | [av]bani : Welcome to America's linguistic future... not the Portugreek of BladeRunner, but it'll do in a pinch ;) |
02:54.38 | [TK]D-Fender | techman97_andy : Same stupid NAT issues? Try a 3rd phone, to see if its just one side thats bitchy. |
02:54.56 | techman97_andy | I'm back - was reading |
02:55.21 | techman97_andy | Fender: I've changed a few things...1 sec |
02:55.32 | cced | who are familiar with zaptel? |
02:56.17 | [TK]D-Fender | cced : just ask the question... |
02:59.12 | tainted- | is DUNDi dead? |
02:59.23 | file[laptop] | okay, DUNDi is a protocol. |
02:59.29 | cced | do zaptel read or write data from hardware using DMA OR mmap? |
02:59.32 | file[laptop] | so it can't exactly be dead |
02:59.35 | justinu | hah |
02:59.37 | tainted- | yes it can |
02:59.39 | justinu | DECnet is dead |
02:59.42 | cced | <[TK]D-Fender> :) |
02:59.53 | justinu | gopher is dead |
02:59.55 | tainted- | smoke signals is a protocol |
02:59.58 | file[laptop] | there's a difference, it's no longer used :P |
03:00.13 | tainted- | well dead = no longer used in a practical sene |
03:00.18 | tainted- | sense |
03:00.20 | *** mode/#asterisk [+o file] by file[laptop] |
03:00.20 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
03:00.21 | justinu | so is latin not a dead language? |
03:00.28 | file[laptop] | but no, some companies are using DUNDi |
03:00.30 | tainted- | do poeple use it |
03:00.40 | tainted- | no one is in #dundi |
03:00.52 | tainted- | is it still actively developed |
03:01.03 | file[laptop] | what else is there to develop on it? |
03:01.11 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
03:01.30 | tainted- | so development is done, shut down irc channel and all support? |
03:01.39 | file[laptop] | did I say that? |
03:01.46 | [TK]D-Fender | cced : no clue... |
03:01.56 | justinu | lol |
03:01.58 | file[laptop] | very few changes are done to DUNDi because it's very complete, it's a finished solution |
03:02.10 | file[laptop] | and companies do use it |
03:02.33 | file[laptop] | and Digium holds no control over who stays in the IRC channel, and who helps with it |
03:02.48 | justinu | well said file |
03:03.03 | cced | <[TK]D-Fender> but I can not find DMA. PCI use mmap |
03:03.28 | file[laptop] | justinu: I try |
03:04.34 | tainted- | file[laptop] so would DUNDi be a good asterisk load balancing solution? |
03:05.07 | orlock | Hmm.. i'm trying to configure a ATA |
03:05.22 | orlock | and i'm not sure what values i should use for RTP base port, or RFC2833 |
03:05.22 | file[laptop] | tainted-: that's not what DUNDi is at it's core |
03:05.27 | orlock | anybody have any suggestions? |
03:05.57 | mog_work | thats why i use it file |
03:05.59 | mog_work | that and failover |
03:06.09 | mog_work | dundi + iax2 +regextern = happiness |
03:06.17 | file[laptop] | that reminds me |
03:06.21 | Cherebrum | orlock: 16384-32767 |
03:06.28 | techman97_andy | WHOOOO HOOOOO! It was just a port thing! I expanded my range and I have full duplex AUDIO! |
03:06.42 | orlock | Cherebrum: for rtp base port? |
03:06.50 | Cherebrum | yes |
03:06.58 | cced | hi :mog_work |
03:07.02 | file[laptop] | mog_work: how was the party? |
03:07.03 | techman97_andy | I have a fully functioning and configured Asterisk system! Thank you to ALL of you who helped me out on this since last Thursday! |
03:07.09 | mog_work | grand |
03:07.11 | techman97_andy | I'm going to freaking bed. |
03:07.12 | techman97_andy | night |
03:07.22 | orlock | cool, what about the rfc2833? |
03:07.25 | orlock | it wants an integer |
03:07.27 | cced | do zaptel read or write data from hardware using DMA OR mmap ? |
03:07.30 | file[laptop] | orlock: 101 |
03:07.39 | cced | mog_work.~ |
03:07.45 | orlock | are you just pulling that out your arse? :) |
03:07.59 | mog_work | i dont think we do either cced |
03:08.04 | file[laptop] | orlock: no that's the dynamic payload that we and every other sane device likes to use for RFC2833 |
03:08.04 | mog_work | but you know where you can find out |
03:08.09 | mog_work | zaptel.c |
03:08.48 | *** join/#asterisk Az_au (i=[+MAL4VO@216.127.73.119) |
03:08.53 | orlock | file[laptop]: cool |
03:09.23 | znoG | i guess you'd need a nice big network of asterisk servers to make use of DUNDi |
03:09.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
03:09.48 | mog_work | i do it with just two znoG |
03:09.53 | mog_work | but the more the merrier |
03:10.00 | znoG | to achieve what, exactly? |
03:10.05 | mog_work | failover |
03:10.08 | cced | mog_work.yes . zaptel provide interface. |
03:10.10 | znoG | i was just reading the DUNDi draft |
03:10.10 | file[laptop] | mog_work is just insane |
03:10.19 | mog_work | its just as good as an iax2 bridge |
03:10.20 | znoG | to understand the protocol a little better |
03:10.27 | mog_work | but it allows for easier growth |
03:10.54 | znoG | and it makes sense to use if you have quite a few asterisk servers, and plan on adding more .. |
03:11.10 | cced | mog_work.yes . tor2.c tor->mem32 = ioremap(tor->xilinx32_region, tor->xilinx32_len); |
03:11.12 | mog_work | it is more beneficial then |
03:11.26 | mog_work | do not look at to drivers |
03:11.45 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
03:11.53 | mog_work | look at cards still sold today |
03:12.13 | justinu|laptop | why aren't tormenta cards sold anymore? |
03:14.26 | mog_work | well digium stopped working on them |
03:14.44 | justinu|laptop | any idea why? it seemed like a cool architecture |
03:14.46 | mog_work | they arent as good as the newer cards from digium/sangoma etc |
03:14.54 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
03:14.58 | mog_work | well digium's hardware is zaptel design |
03:15.00 | mog_work | just not a tor |
03:15.07 | mog_work | tors also where not as scalable |
03:15.11 | cced | yes: most card use PCI slot.yes PCI |
03:15.16 | justinu|laptop | you know any specifics? |
03:15.25 | mog_work | i could go into it |
03:15.32 | mog_work | but i only know what was told to me |
03:15.34 | justinu|laptop | if you're up to it |
03:15.42 | mog_work | i know that driving fulll 4 e1s on a tor2 is very flakey |
03:15.54 | justinu|laptop | too many interupts/sec or something? |
03:16.12 | cced | yes. vhdl problem.. :) |
03:16.32 | justinu|laptop | cced: are you a pot? |
03:16.44 | justinu|laptop | s/pot/bot/ |
03:16.50 | justinu|laptop | a bot on pot? |
03:18.44 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
03:19.53 | asterboy | did someone say pot? |
03:20.00 | asterboy | high |
03:20.14 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
03:20.25 | cced2 | <cced> pot ? what mean? |
03:21.27 | justinu|laptop | haha |
03:21.50 | justinu|laptop | cced2: what is your native language? |
03:22.07 | cced2 | Chinese |
03:23.43 | cced2 | what mean?puzzle |
03:25.04 | *** join/#asterisk Redimido (n=gorv@201.153.212.239) |
03:25.11 | cced2 | mog_work : in libpri ,where define t203 =10s ? |
03:25.47 | mog_work | cced2, i dont know how to explain this politely |
03:25.50 | mog_work | im happy to help |
03:25.54 | mog_work | but i am not google |
03:25.56 | mog_work | or the source |
03:26.04 | mog_work | both of those you have |
03:26.07 | mog_work | use them |
03:26.13 | mog_work | if you have real questions |
03:26.15 | mog_work | ask me |
03:28.19 | file[laptop] | ha |
03:28.21 | harryvv | google makes a google box |
03:28.53 | harryvv | file, whats up? |
03:29.07 | harryvv | cced cantonese? |
03:29.07 | file[laptop] | sitting in bed contemplating sleep |
03:29.16 | orlock | man, we are trying to configure a netcomm ata |
03:29.19 | asterboy | [park] on the Polycom Phone seems to want to accept some digits and then you press [park] again but it does nothing. What does work to park a call is to [Transfer] to a parkinglot and then put the call on hold for pickup at another phone. Anyone know if the Polycom Park is suppoerted and if so what the configuration parameters are? |
03:29.19 | orlock | its not fun |
03:30.52 | asterboy | Anyone here used the Polycom [Park] feature button? |
03:31.14 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
03:31.33 | cced2 | mog_work: o thanks . I read codes :) |
03:33.18 | asterboy | Anyone use the [Services] button on a Polycom IP600? |
03:33.27 | [TK]D-Fender | I do |
03:33.30 | asterboy | Looks like a microbrowser |
03:33.38 | [TK]D-Fender | thats exactly what it is. |
03:34.33 | asterboy | I thought it was only supported in the IP601 and 501 series. |
03:34.43 | [TK]D-Fender | nope, only on 60x |
03:36.47 | asterboy | hmmm...time to learn up on what that can do. |
03:37.17 | asterboy | http://www.voip-info.org/wiki/view/Polycom+Microbrowser |
03:37.21 | [TK]D-Fender | asterboy : not that much. no control over phone functions yet |
03:37.46 | asterboy | "There is very limited information about this feature in the Polycom documentation. " |
03:38.21 | asterboy | no doubt, seems to be tantalizing though |
03:38.27 | *** part/#asterisk Redimido (n=gorv@201.153.212.239) |
03:38.27 | [TK]D-Fender | You can use it to display live infor on "idle" and provide basic HTML (VERY) as interactive pages to do something else "useful" |
03:38.50 | [TK]D-Fender | asterboy : the A2H handbook pages ont he WIKI have a lot more detail stangely... |
03:38.56 | [TK]D-Fender | A@H |
03:39.24 | asterboy | Looks like it would be good for a business card of the business. |
03:39.33 | asterboy | To show address and stuff for temp workers. |
03:39.35 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
03:40.11 | asterboy | wow ... even inline images |
03:41.19 | asterboy | I might put my business card in there for when the client needs support. |
03:42.21 | [TK]D-Fender | I use it for mass-presence checking, company contact list and live queue stats |
03:42.37 | asterboy | sweet |
03:42.44 | asterboy | live queue stat |
03:43.10 | asterboy | mass-presence...is that like FOP? |
03:44.11 | asterboy | "Main Browser Home. |
03:44.16 | asterboy | Where is that set? |
03:44.50 | [TK]D-Fender | asterboy : something like that. I get "ext - name" for each phone in use |
03:45.15 | [TK]D-Fender | asterboy typically in your provisioning files. |
03:45.20 | asterboy | That kicks ass. |
03:45.38 | [TK]D-Fender | I do it in phonexxx.cfg so as to personalize it per phone |
03:45.42 | asterboy | Like <mac>-phone.cfg? |
03:45.46 | asterboy | ah |
03:46.02 | [TK]D-Fender | no, in the one referenced by <mac>.cfg |
03:46.11 | asterboy | ok |
03:46.37 | *** join/#asterisk hatamen (n=hatamen@222.183.27.190) |
03:47.04 | asterboy | interesting...nothing in the standard .cfg file. |
03:47.40 | [TK]D-Fender | asterboy : nope, I copied it from sip.cfg. you can port most of the XML tree fromt he master to overrider whatever you need on a per-phone basis |
03:47.54 | asterboy | ah |
03:49.08 | asterboy | so something like <main mb.main.home="http://domain.com/polycom.html"/> |
03:49.17 | asterboy | and then put in supported tags. |
03:49.40 | [TK]D-Fender | yup, just a few basic ones needed. |
03:50.30 | [TK]D-Fender | I've done forms on it, minimal font control (avoid where possible), inline images (my queue stats idle page has out company logo in it and info on 2 queues & 2 VM boxes |
03:50.47 | asterboy | nice |
03:51.15 | asterboy | I like the logo idea...can that be put on the front default display with the clock? |
03:51.15 | [TK]D-Fender | And all ring like Cisco's on "24" :D |
03:51.33 | asterboy | that would be creepy. |
03:51.46 | [TK]D-Fender | asterboy : 2 ways on 601. static image or idle XHTML on time interval. |
03:52.22 | asterboy | mb.idleDisplay.home="(url)" and mb.idleDisplay.refresh="(seconds"). |
03:52.29 | [TK]D-Fender | yup |
03:53.06 | [TK]D-Fender | very usable... thats how I've got "live queue" stats. every 10 sec I call a PHP'd page that uses AMI to poll queues & VMboxes |
03:53.10 | asterboy | only 4bpp bmp support though |
03:53.31 | [TK]D-Fender | asterboy I believe 8 bit works and is scaled |
03:53.39 | file[laptop] | commit commit here, commit commit there |
03:53.42 | asterboy | that would be better |
03:54.21 | file[laptop] | eep |
03:55.02 | [TK]D-Fender | You know with just a speakerphone, the IP301 would totally rock.... |
03:55.20 | [TK]D-Fender | And my 501 is still on backorder... dammit |
03:55.49 | k-man | is there a tapi driver that can communicate with an ip phone |
03:55.58 | asterboy | I'm totally happy with my IP600s |
03:56.20 | asterboy | Everything works, save [Park] & IM |
03:56.20 | [TK]D-Fender | asterboy : as am I with mine.... I jsut want one of each at home.... |
03:56.32 | tainted- | anyone work with Vovida's load balancer proxy? |
03:56.36 | asterboy | lol...ya I have the IP500 and IP300 |
03:56.49 | asterboy | eBay items |
03:57.10 | asterboy | th 01s would be nice for the bigger memory though |
03:57.20 | [TK]D-Fender | I boght a 301 & 501 new, billed to my consultancy so hopefully I can write them off as "training tools" |
03:57.38 | asterboy | totally write offable. |
03:58.16 | asterboy | I want to switch from my computer consultancy to telecom consultancy. |
03:58.28 | asterboy | Just enjoy it so much more. |
03:58.59 | asterboy | The telecom company is just starting to take off. |
04:00.10 | [hC] | I wanna know what its gonna take to get more than 7 line statuses working on the polycoms |
04:00.23 | asterboy | v1.4 when it comes out |
04:00.26 | [hC] | Why has nobody implemented the secondary BLF protocol that polycom doesnt have an issue with? |
04:00.28 | asterboy | or SIP-B |
04:00.40 | [hC] | SIP-B? I wasnt sure what it was called. |
04:00.42 | file[laptop] | [hC]: implement it then. |
04:00.52 | asterboy | SIP for Business |
04:00.53 | [hC] | Im not a coder or i would have done it already :S |
04:00.58 | [hC] | I'm a network engineer/integrator |
04:01.13 | asterboy | Who needs it when you can use the microbrowser anyway. |
04:01.23 | xachen | i can do perl and php ^_^ |
04:01.24 | [hC] | My clients receptionists |
04:01.26 | xachen | just not real good C |
04:01.53 | asterboy | I can program in anything...when getting paid. |
04:02.19 | [hC] | So, any eta on firmware 1.4? or maybe, has anyone attempted to implement SIP-B? I'm more than willing to bounty it, if its worth it |
04:02.24 | asterboy | otherwise...I just live at the bash prompt |
04:02.26 | [TK]D-Fender | I have XHTML MicroBroser stuff to compesate for "live" presence suppotr" |
04:02.41 | [hC] | [TK]D-Fender: willing to share it? :) |
04:03.05 | asterboy | ya, that's what I'm saying [TK]...the microbrowser takes care of that. |
04:03.06 | [TK]D-Fender | [hC] : Polycom is supposed to be effectively removing their artificial limi in the next SIP release |
04:03.19 | [TK]D-Fender | [hC] : when I'm at work tomorrow, sure. |
04:03.32 | [hC] | [TK]D-Fender: yeah... any idea when they plan on releasing the new firmware? I mean, have they actually made a statement as to when? |
04:03.58 | [TK]D-Fender | [hC] : "When its done" ;p |
04:04.53 | file[laptop] | good answer |
04:04.56 | [hC] | [TK]D-Fender: Figures :) |
04:08.37 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
04:12.26 | *** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net) |
04:14.19 | [hC] | Ive still not touched hte microbrowser at all |
04:14.27 | asterboy | Just received my box of Grandstream GXP-2000s |
04:14.48 | tainted- | good now box them back up and buy some real phones |
04:14.59 | asterboy | lol |
04:15.01 | tainted- | ;) |
04:15.19 | asterboy | I tried to sell the Polycoms but the client was cheapo |
04:15.42 | asterboy | Didn't want to loose the bid, so I caved in. |
04:16.02 | asterboy | Now I get to try out these phones and see if they live up to the garbage name they have. |
04:16.16 | asterboy | Hopefully they are at least functional. |
04:17.13 | asterboy | [hC], I just touched the microbrowser for the first time...and ah...well the page is blank. |
04:17.41 | asterboy | back to the playing with the settings...not sure what I did wrong...left only the simple tags. |
04:18.31 | orlock | Hmm, has anybody here used a netcomm ata? |
04:18.48 | [TK]D-Fender | asterboy : you need to put in the basics, even if empty : HTML, HEAD, TITLE, BOD, and make sure everything you do has a closing tag |
04:19.45 | asterboy | got it...tried to get fancy and put a bgcolor field in the body tag...it don't like it. |
04:20.32 | asterboy | Thanks Polycom you don't have to reboot phone after html changes. |
04:20.51 | bkw_ | http://www.ngnsky.com/product_info.php?products_id=49 |
04:21.07 | justinu|laptop | asterboy: you're gonna need a lot of weed if you're trying to work with gxp2000's |
04:21.18 | asterboy | LOL |
04:21.29 | asterboy | I was just going to twist one before starting. |
04:22.01 | Qwell | WTF |
04:22.05 | Qwell | bkw_: That site is crap |
04:22.12 | justinu|laptop | they're fine for occasional users |
04:22.20 | asterboy | microbrowser does not like <br> |
04:22.27 | justinu|laptop | but for business customers... i dunno |
04:22.27 | asterboy | wants <br/> |
04:22.30 | Qwell | "The version of your browser is not supported. Please use InternetExplorer version 4 and above!" |
04:22.43 | Qwell | site looks...fine...in mozilla |
04:22.58 | asterboy | Dillo will break it. |
04:23.24 | Qwell | asterboy: <br> isn't valid.. |
04:23.32 | Qwell | <br/> or <br></br> are |
04:23.45 | asterboy | ya, it sure lets you know. |
04:24.10 | asterboy | I'm thinking CSS is not supported either :P |
04:25.35 | OliverX | good morning asterisk world (; |
04:26.07 | xachen | bkw_: thats way cheaper than digium :o |
04:26.24 | [TK]D-Fender | asterboy : Like I said, EVERYTHING has to be closed.. that'd be <br /> |
04:27.11 | OliverX | is their a interface/api to use mysql with asterisk? |
04:27.30 | *** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net) |
04:27.41 | drfoomod2 | has anyone seen a asterisk@home box hang at Checking for new hardware? |
04:27.43 | asterboy | yes, it sure complains if you get anyting wrong. Kinda like programming in Pascal |
04:28.11 | drfoomod2 | this is the first boot after installing 2.7 |
04:28.25 | Qwell | drfoomod2: #asteriskathome |
04:28.30 | drfoomod2 | tx :) |
04:28.49 | justinu|laptop | anyone know how this works? |
04:28.52 | justinu|laptop | http://en.wikipedia.org/wiki/Unlicensed_Mobile_Access |
04:29.17 | drfoomod2 | Qwell: empty chan |
04:29.20 | [hC] | f this im gonna go home |
04:29.28 | Qwell | [hC]: and test my patch? :D |
04:29.33 | [hC] | drfoomod2: they moved, #freeswitch |
04:29.39 | asterboy | home sweet home |
04:29.48 | [hC] | Qwell: yep :) |
04:29.52 | asterboy | [hC] come on over and twist one with me. |
04:29.54 | Qwell | excellent... |
04:30.01 | Qwell | Just make sure to test an addon |
04:30.09 | [hC] | the 7914 is attached |
04:30.09 | justinu|laptop | asterboy: where you live? |
04:30.11 | [hC] | where is the patch again? |
04:30.15 | OliverX | is their an answer for me? |
04:30.16 | asterboy | Alberta bound |
04:30.20 | Qwell | 6859 |
04:30.22 | justinu|laptop | ah |
04:30.22 | asterboy | land locked in Canada eh! |
04:30.29 | [hC] | eek that box is running ast 1.2.1 |
04:30.32 | [hC] | will it compile against that? |
04:30.35 | Qwell | no, heh |
04:30.37 | Qwell | trunk only |
04:30.37 | justinu|laptop | i live in the wasteland called los angeles |
04:30.38 | asterboy | I feast on whale blubber and live in the snow |
04:30.42 | [hC] | hmm |
04:30.43 | [hC] | okay |
04:30.45 | asterboy | lost angles |
04:30.46 | [hC] | might take some time then |
04:30.51 | [hC] | that thing is a production box |
04:30.52 | [hC] | :) |
04:30.53 | asterboy | been there at night |
04:30.56 | Qwell | eww |
04:31.02 | asterboy | not even the cops will help you out. |
04:31.02 | [hC] | i know |
04:31.03 | [TK]D-Fender | Pascal = teh shiznit yo! |
04:31.12 | Qwell | [hC]: It'll crash a lot, so...not recommended in production :p |
04:31.14 | xachen | asterboy: you from Alberta? :O |
04:31.19 | OliverX | hm i must go to work. perhaps i became an answer in this evening(german time +1) |
04:31.23 | [hC] | I'll try it on my astlinux box |
04:31.30 | [hC] | i'll see if that thing will let me upgrade to trunk easily ;) |
04:31.30 | asterboy | yepper...I'm as red necked as an Albertan can get. |
04:31.40 | drfoomod2 | [hC]: tx |
04:31.42 | justinu|laptop | cops are criminals w/ badges |
04:31.45 | [hC] | <- vancouver |
04:31.58 | [TK]D-Fender | OMG |
04:32.05 | asterboy | justinu, you certainly have that close to the mark...not all, but most. |
04:32.09 | [hC] | WTFLOLBBQ? |
04:32.17 | [hC] | :) |
04:32.21 | asterboy | Alberta is a Police State |
04:32.22 | [hC] | sorry bkw |
04:32.25 | bkw_ | haha |
04:32.34 | [hC] | I couldnt RESIST |
04:32.44 | bkw_ | they moved to #freepbx |
04:32.49 | xachen | asterboy: me too :p |
04:32.53 | [hC] | freepbx, freeswitch... so close :P |
04:33.03 | justinu|laptop | why is it a police state? |
04:33.10 | asterboy | xachen, Alberta is smoking for business right now. |
04:33.20 | xachen | yup |
04:33.46 | alephcom | I don't quite see how that makes it a police state but maybe that's cause I'm way out in the country. |
04:33.53 | [hC] | Hmm.. I need a LAMP programmer to make a front end for my stuff, anyone know of anyone good in vancouver? or even remote as long as they're really good :) |
04:34.13 | xachen | hehe |
04:34.14 | [hC] | Since all the asterisk front end config suites are crap, or force you to do things their way |
04:34.17 | xachen | aleph is way out in the middle of nowhere |
04:34.23 | asterboy | justinu, because the cops here micro manage everything you do. |
04:34.46 | SplasPood | [hC]: heh.. |
04:35.05 | alephcom | asterboy: What have you done that they watch you? :-) |
04:35.16 | asterboy | justinnu, I've had them come to my door for my dog being out front, drumming at 9pm, shovling snow too close to the sidewalk...etc etc etc. |
04:35.37 | alephcom | Calgary? Edmonton? |
04:35.50 | asterboy | alephcom, they watch everyone like a hawk here. |
04:36.16 | asterboy | They even arrested an RCMP officer because for gun possesion. |
04:36.17 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
04:36.17 | *** mode/#asterisk [+o denon] by ChanServ |
04:37.02 | alephcom | xachen is right that I'm in the middle of nowhere. Let's see. I saw an RCMP truck on my road 1.5 years ago. lol |
04:37.08 | xachen | hehe |
04:37.11 | xachen | about a year here |
04:37.22 | [hC] | SplasPood: hey |
04:37.30 | asterboy | Ya, I need to move out into the country. |
04:37.36 | asterboy | That sounds peacful. |
04:37.41 | [hC] | SplasPood: id contract you if you want, but i need something done in like... under 2 months, and you have a full time job already :) |
04:37.52 | asterboy | Do you have HighSpeed Internet though? |
04:38.11 | alephcom | Yeah, of course. |
04:38.25 | asterboy | can't be too far from city central |
04:38.30 | alephcom | Wireless with decent speeds. It's not a cheap as adsl. |
04:38.37 | xachen | im getting wycom :( |
04:38.42 | alephcom | :-( |
04:38.46 | xachen | should be in by the end of the week |
04:38.47 | asterboy | Ya, there is a lot of WISP going on in Alberta now. |
04:39.22 | asterboy | Alberta oil sands just got estimated at 1.5 Trillion Dollars. |
04:39.44 | asterboy | Alberta is about to become the World's #1 Oil producer. |
04:39.50 | [hC] | Im going to send them an email and ask for a pizza, since they can obviously afford it. |
04:40.21 | *** join/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
04:40.30 | asterboy | I was thinking of going up to Fort McMurray to sell phone systems to the Oil patch. |
04:40.56 | xachen | make them newfie proof first.... |
04:40.57 | xachen | big numbers |
04:41.01 | alephcom | It's rather sad that Klein is quitting. Darren ducks |
04:41.03 | xachen | and a big green and red button :p |
04:41.11 | xachen | and you suck |
04:41.23 | xachen | Klein can go $#$! himself |
04:41.47 | asterboy | lol |
04:42.08 | drfoomod2 | can i use a cisco router (such as a 2621xm) with a t1 card to access FXS ports on an adtran channel bank? |
04:42.31 | asterboy | I laughed when Klein went down to the homeless shelter, drunk, and threw money at them and told them to back to work. |
04:42.43 | xachen | i'm going to sleep |
04:43.06 | asterboy | xachen, keep in touch on here. |
04:43.45 | asterboy | The project head for LinuxFromScratch lives in Canmore, Alberta. |
04:43.58 | xachen | yeh |
04:44.01 | xachen | OpenBSD in Calgary |
04:44.07 | xachen | but I'm out |
04:44.09 | [hC] | Did you guys know Erik Reid? |
04:44.11 | asterboy | night |
04:44.30 | asterboy | Erik Reid, no...who is he? |
04:44.36 | [TK]D-Fender | asterboy : yeah, but can his being that far away from "civilization" truely be called "life" ;) |
04:44.48 | [hC] | former open/netbsd guy who lives in alberta |
04:44.53 | asterboy | lol...seriously, Canmore rocks. |
04:44.55 | [hC] | he died a couple years ago though |
04:45.15 | asterboy | what was his function in the hive? |
04:45.32 | [TK]D-Fender | ok, I'm out... later all |
04:45.40 | asterboy | night TK |
04:45.53 | asterboy | thanks for the info the microbrowser |
04:48.21 | kimosabe | i have a sip acount i used to use it with a sipura device but now i have a tdm400p with ine fxs on my asterisk box does any one have a config example so that my fxs card will use that sip acount |
04:48.26 | *** join/#asterisk [Mojo] (n=mojito@200-122-80-171.cab.prima.net.ar) |
04:51.04 | asterboy | kimosabe, its much the same setup save that you can tell your FXS extension to Dial(SIP |
04:51.18 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
04:51.23 | xbmodder_lappy | I keep getting this error: |
04:51.23 | xbmodder_lappy | == Spawn extension (pushup, 19252097312, 3) exited non-zero on 'SIP/pushup-608e' |
04:51.48 | kimosabe | asterboy do you know where i can get an exampl epleazse |
04:52.54 | asterboy | looking. |
04:53.12 | kimosabe | thanks man |
04:53.16 | asterboy | I just set this up for a client so I'll see if I can cut and paste. |
04:55.21 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
04:55.29 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
05:03.39 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
05:11.05 | asterboy | kimosabe, here is your SIP to FXS example config: http://pastebin.ca/48975 |
05:11.46 | asterboy | I setup G729, had to buy a license from Digium...so setup your codec as required. |
05:14.31 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
05:18.08 | *** join/#asterisk kisu (n=daniel@cielkisu.tb.as8758.net) |
05:18.14 | asterboy | kimosabe, Here is a better version with more contexts: http://pastebin.ca/48976 |
05:18.22 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
05:21.50 | asterboy | Nice! Check this out: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf |
05:23.17 | asterboy | jbot, refcard is a nice printout cheet sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf |
05:23.18 | jbot | ...but refcard is already something else... |
05:23.23 | asterboy | ~refcard |
05:23.25 | jbot | extra, extra, read all about it, refcard is http://people.debian.org/~debacle/refcard/ |
05:24.05 | asterboy | jbot, refcard is also a nice printout cheet sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf |
05:24.07 | jbot | okay, asterboy |
05:24.11 | asterboy | ~refcard |
05:24.13 | jbot | [refcard] http://people.debian.org/~debacle/refcard/, or a nice printout cheet sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf |
05:24.27 | asterboy | good boy. |
05:25.12 | Qwell | jbot: no, refcard is http://people.debian.org/~debacle/refcard/, or a nice printout cheat sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf |
05:25.13 | jbot | Qwell: okay |
05:25.16 | Qwell | :D |
05:25.17 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
05:25.38 | asterboy | :P |
05:25.43 | FuriousGeorge | hey all |
05:26.13 | asterboy | Furious, howdy ho. |
05:27.33 | FuriousGeorge | how much does someone who installs a 10-20 extension pbx charge for basic service: tech support 9-5, next day repair replacement, etc |
05:28.18 | FuriousGeorge | well assume unlimited phone support 9-5 |
05:28.40 | asterboy | jbot, refcard is also a vi card here: http://tnerual.eriogerg.free.fr/vimqrc.pdf |
05:28.41 | jbot | okay, asterboy |
05:28.47 | asterboy | ~refcard |
05:28.48 | jbot | i guess refcard is http://people.debian.org/~debacle/refcard/, or a nice printout cheat sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf, or a vi card here: http://tnerual.eriogerg.free.fr/vimqrc.pdf |
05:29.25 | asterboy | vi rulez! |
05:29.47 | FuriousGeorge | asterboy: can you explain something to me |
05:30.26 | *** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net) |
05:30.38 | Supaplex | ;) |
05:30.41 | asterboy | lol |
05:31.08 | FuriousGeorge | lol, im not trying to be flip, but from the little ive used vi, even when you got the keystrokes in muscle memory, why would you want to hit all those buttons to edit an asterisk conf? |
05:31.16 | FuriousGeorge | keep in mind i only use nano :) |
05:31.39 | FuriousGeorge | not emacs, i would ask the same to an emacs user |
05:31.40 | Az_au | when you know all the shortcuts it cuts down on keystrokes ;) |
05:31.45 | asterboy | I'm old school...been using Xenix since late 80's |
05:31.56 | asterboy | I really don't like GUI |
05:32.00 | FuriousGeorge | ~xenix |
05:32.02 | jbot | i heard xenix is NOT that bad.. not that it's that good. or a Unix derivate by M$ aborted to focus on NT. MS sold it to SCO who made SCO Xenix which became SCO Unix |
05:32.27 | asterboy | ~MAI |
05:32.45 | Supaplex | well, vi/vim/etc doesn't require most of the keyboard to function :) |
05:33.29 | FuriousGeorge | ive been planning to learn some C and PHP so i guess i find out for myself |
05:33.53 | Az_au | ya.. syntax hilighting is also a bonus |
05:34.02 | asterboy | man I must be old, no MAI |
05:34.16 | Az_au | and folds |
05:34.19 | FuriousGeorge | asterboy: tell jbot what mai is |
05:34.26 | FuriousGeorge | so that i'll know too :) |
05:36.10 | Supaplex | and you won't find emacs on an embeded system :P |
05:36.28 | FuriousGeorge | are learning logical languages like spoken languages in that they suggest you learn one at a time so you dont confuse yourself? |
05:36.31 | Qwell | Supaplex: But you'll find ed! |
05:36.32 | asterboy | jbot, MAI is Management Assistance Inc. which asterboy first cut his programming teeth on. More info here: http://web.archive.org/web/20050305205751/www.science.uva.nl/museum/basicfour_tbl.html |
05:36.33 | jbot | asterboy: okay |
05:36.35 | mog_work | i thought emacs was an embedded system Supaplex |
05:37.54 | Supaplex | real sysadmins use ed |
05:38.04 | Qwell | ed is the standard text editor! |
05:38.05 | mog_work | bump ed |
05:38.12 | mog_work | dump text into device buffer |
05:38.17 | Supaplex | cat! |
05:38.19 | Az_au | ya.. cat |
05:38.21 | mog_work | you should know evreything on box |
05:38.26 | mog_work | so no need to read it back out |
05:38.30 | Qwell | cat? pfft, newbs |
05:38.31 | Supaplex | or echo :) |
05:38.37 | Qwell | < and > |
05:38.48 | mog_work | amen brother Qwell |
05:38.53 | Qwell | korn has some nifty stuff for redirection |
05:39.04 | Qwell | ksh |
05:39.39 | *** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net) |
05:40.23 | justinu | is xenix what made you start smoking pot in the first place? |
05:40.29 | asterboy | I really love this site: http://www.old-computers.com/ |
05:40.44 | FuriousGeorge | ~jbot |
05:40.45 | jbot | jbot is, like, only marginally useful at best, He got a C- on his Turing Test |
05:40.46 | asterboy | lol, no...my wife started me smoking pot. |
05:40.53 | justinu | lol thats funny |
05:40.54 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
05:40.59 | justinu | but understandasble |
05:41.23 | FuriousGeorge | asterboy: still married to her |
05:41.31 | FuriousGeorge | ? |
05:41.41 | asterboy | oh ya...the only women who could put up with a guy like me. |
05:41.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
05:41.49 | FuriousGeorge | lol |
05:42.11 | justinu | i'm getting married next monday |
05:42.25 | asterboy | Although I'm not gay, I can see why some men go brokeback. |
05:42.38 | asterboy | justinu, seriously? |
05:42.40 | justinu | yes |
05:43.01 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
05:43.03 | asterboy | don't know if I should congradulate you or feel sorry for you. |
05:43.06 | FuriousGeorge | name the artist: i was in love with a girl on marijuana. she said "if i'm not stoned, i don't wanna." But she got so paranoid, her place i would avoid. (i was in love with a girl on marijuana |
05:43.08 | justinu | going to NYC tomorow morning for my last trip as a single man |
05:43.14 | asterboy | never mind...misery enjoys company. |
05:44.10 | asterboy | Tommy? |
05:44.17 | FuriousGeorge | yeah |
05:44.31 | FuriousGeorge | Tom Petty "Girl on LSD" really funny song |
05:44.39 | FuriousGeorge | haha you knew that. stoner |
05:44.43 | FuriousGeorge | :) |
05:44.51 | justinu | ~asterboy |
05:44.52 | jbot | [asterboy] a weed smoker |
05:45.07 | FuriousGeorge | it could be worse |
05:45.11 | FuriousGeorge | ~furiousgeorge |
05:45.13 | jbot | from memory, furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
05:45.14 | justinu | yes, far worse |
05:45.24 | justinu | where is the man with the yellow bat, anways? |
05:45.31 | asterboy | lol |
05:45.38 | FuriousGeorge | he's about |
05:46.00 | FuriousGeorge | he usually tries to sneak up on me so he changes his nick right before the attack |
05:46.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:46.33 | justinu | when i hear man with yellow bat, i think of fooly cooly |
05:46.43 | justinu | but that was something like a hockey stick, i think |
05:47.05 | asterboy | lol, I had one of these: http://www.old-computers.com/museum/computer.asp?st=1&c=1170 |
05:47.41 | FuriousGeorge | why is it that i never get tired of seeing my name in jbot |
05:47.54 | asterboy | and one of these: http://www.old-computers.com/museum/computer.asp?c=477&st=1 |
05:48.16 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
05:48.27 | asterboy | and this: http://www.old-computers.com/museum/computer.asp?c=102&st=1 |
05:48.31 | FuriousGeorge | asterboy: how did you play doom III on that! |
05:48.43 | asterboy | lol |
05:49.18 | asterboy | man I'm not even 40 and I feel old |
05:50.13 | Qwell | asterboy: That museum sucks... |
05:50.20 | FuriousGeorge | lol |
05:50.22 | Qwell | Doesn't even have my zenith |
05:51.09 | FuriousGeorge | i had an intelivision but i think that one was already a little pop-culture |
05:51.34 | asterboy | lol...no zenith |
05:52.07 | asterboy | I remember beggin for an intelivision |
05:52.37 | FuriousGeorge | so how much is your average basic/silver/whatever service agreement gonna run monthly for 10 -20 extensions? |
05:52.41 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
05:53.03 | X-Rob | Oooh, intellivisons were so cool. you could get a keyboard and it was just like a computer! |
05:53.20 | FuriousGeorge | X-Rob: i just remember playing baseball |
05:53.31 | FuriousGeorge | i was 4 or so |
05:54.28 | litage | are there any devices (other than the TDM400P and similar PCI cards) that have 2 FXS and 2 FXO ports? |
05:54.51 | FuriousGeorge | litage: does the sangoma card do 2X2? |
05:54.56 | FuriousGeorge | or is it 4 per? |
05:55.05 | asterboy | now I have 3 netfinities and 3 LH 6000 HP NetServers pilled up doing nothing. |
05:55.13 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
05:55.16 | asterboy | useless. |
05:55.25 | *** join/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net) |
05:55.39 | asterboy | litage, not sure if the sangoma has that. |
05:56.22 | asterboy | sagnoma |
05:57.20 | MikeJ[Laptop] | sangoma's analog is based on 2 port mods |
05:57.28 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
05:57.30 | MikeJ[Laptop] | up to 24 ports |
05:57.45 | litage | FuriousGeorge, asterboy: if it does or doesn't, i'm looking for non-PCI card hardware |
05:58.06 | X-Rob | litage, usb? |
05:58.12 | FuriousGeorge | T1? |
05:58.25 | FuriousGeorge | you can get 2 port ATAs from linksys |
05:58.25 | Telamon | Is there a variable which gives the login ID (ie, the SIP or IAX2 username)? |
05:58.51 | FuriousGeorge | and some sip -> pots things i think theyre called iaxy |
05:59.12 | asterboy | mediatrix and quitum make some nice gates |
05:59.27 | asterboy | s/qui/quin/ |
05:59.52 | SwK | screw mediatrix and quintums |
05:59.53 | litage | FuriousGeorge: none of linksys' products support 2 FXOs |
05:59.55 | MikeJ[Laptop] | I think for low qantity fxs, sipura is still the best cost/port isnt it? |
06:00.00 | SwK | get a Audiocodes |
06:00.13 | MikeJ[Laptop] | SwK, what's cost/port? |
06:00.18 | SwK | if you need more then 2FXS |
06:00.51 | asterboy | For 2FXS the cheapest is to get an unlocked Linksys PAP2 |
06:01.03 | SwK | they arent cheap but they are worth it... 729, 723, T38 etc |
06:01.08 | asterboy | but they are a big pain in the ass |
06:01.12 | *** join/#asterisk BugKham (n=lamer@ppp-58.8.4.140.revip2.asianet.co.th) |
06:01.16 | litage | asterboy: yes, but that doesn't have any FXO ports |
06:01.20 | asterboy | true |
06:01.30 | SwK | litage: you need mixed FXO/FXS device? |
06:01.47 | litage | SwK: yes, i need a device with 2 FXO and 2 FXS ports |
06:02.03 | SwK | litage: only way to get that is TDM400 |
06:02.05 | asterboy | not cheap: http://cgi.ebay.com/Quintum-Tenor-ASM200-2FXS-2FXO-VoIP-H323-SIP-Gateway_W0QQitemZ5750195904QQcategoryZ61839QQssPageNameZWD1VQQrdZ1QQcmdZViewItem |
06:02.18 | SwK | quintums are a POS |
06:02.29 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
06:02.29 | *** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com) |
06:02.38 | FuriousGeorge | litage: i guess it was just a similar color to a linksys pap2 |
06:02.47 | FuriousGeorge | anyway, i know ive seenem |
06:02.48 | asterboy | ya, I'd much rather go with TDM |
06:03.14 | SwK | if you gave me a Quintum, I would ebay it and buy an Audiocodes heh |
06:03.52 | asterboy | http://cgi.ebay.com/Cisco-VOIP-VG200-Router-NM-2V-with-VIC-2FXO-2FXS_W0QQitemZ9708191990QQcategoryZ51204QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
06:04.38 | asterboy | Anyone use Micronet? |
06:04.41 | asterboy | http://cgi.ebay.com/2-Micronet-SP5014-VoIP-Routers-2FXO-2FXS-ports_W0QQitemZ9708613398QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
06:05.01 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
06:05.11 | asterboy | Think they are made in New Zealand. |
06:05.15 | asterboy | iirc |
06:05.35 | MikeJ[Laptop] | what? newzelanders? |
06:06.11 | MikeJ[Laptop] | oh oh SwK!! |
06:06.55 | SwK | ? |
06:07.19 | MikeJ[Laptop] | wassup man |
06:07.40 | SwK | nadda |
06:07.40 | MikeJ[Laptop] | I want to publically announce btw to never ever use interland |
06:07.57 | SwK | hah |
06:08.23 | austinnichols101 | interland sux0rs |
06:08.36 | MikeJ[Laptop] | indeed |
06:08.40 | SwK | I would also like to publically say Never use Nocster.com aka burst.net they are the suck |
06:08.45 | MikeJ[Laptop] | 16hrs to properly escalate a 5 min fix |
06:08.58 | SwK | hah |
06:09.05 | MikeJ[Laptop] | the whole time fighting with me about it being my prob |
06:09.14 | BugKham | Hi there, what's the use of the 'username' parameter in sip.conf? in my client I only need to put the 'section title' |
06:09.31 | MikeJ[Laptop] | every time, I carefully explain (clearly over their head) how the problem was their issue in great technical detail |
06:09.33 | X-Rob | BugKham, I think you're confusing Microsoft Word with a SIP phone. |
06:09.41 | SwK | try 2 days down because they decided my server needed to have the OS reinstalled and then couldnt tell me why no one could connect to the server |
06:09.52 | MikeJ[Laptop] | they say they escalated it and it took 8+hrs to get tot hte top of the queue |
06:09.53 | BugKham | for both username and auth username |
06:09.58 | asterboy | That sounds like an offshore India tech support scenario |
06:10.07 | MikeJ[Laptop] | in reality... no one bothered till I was a total jerk |
06:10.17 | X-Rob | MikeJ[Laptop], and you wonder why I' |
06:10.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
06:10.22 | X-Rob | I've got my own hardware? |
06:10.28 | MikeJ[Laptop] | asterboy, there are plenty of offshores that do a good job |
06:10.34 | MikeJ[Laptop] | me too... |
06:10.41 | asterboy | have yet to find one |
06:10.44 | X-Rob | http://www.serverpronto.com |
06:10.44 | MikeJ[Laptop] | but this was a side deal for my dads office |
06:10.46 | BugKham | X-Rob: huh, my sip client does care about my username parameter |
06:10.49 | MikeJ[Laptop] | I just took it over |
06:10.57 | FuriousGeorge | im installing a phone system for a small business (17 users) and i dont wanna offer them a service agreement, anyone in here wanna do it? |
06:11.02 | MikeJ[Laptop] | so it basically runs itself till somthing like this blows up |
06:11.10 | BugKham | X-Rob: the section title is all it needs |
06:11.15 | SwK | you dont have it hosted where you work mikej? |
06:11.18 | asterboy | Furious, where? |
06:11.25 | litage | the Audiocodes MP-20x datasheet (http://www.audiocodes.com/Objects/LTRM_30008_DS_MP-20x.pdf) says it connects "2 POTS lines or fax machines". how can an RJ-11 port be both an FXO and FXS port? |
06:11.26 | FuriousGeorge | central nj |
06:11.35 | FuriousGeorge | northern central kinda east :) |
06:11.36 | SwK | Furiousgeorge: remote support ok? |
06:11.48 | FuriousGeorge | SwK sure till the PSU fires |
06:11.51 | asterboy | I'm out...I'd have to get a green card. |
06:11.51 | FuriousGeorge | fries** |
06:11.58 | FuriousGeorge | lol |
06:12.10 | asterboy | once the smell pot on me at the airport...I'm done. |
06:12.24 | FuriousGeorge | asterboy: or is it hash accross the pond? |
06:12.32 | asterboy | lol |
06:12.33 | austinnichols101 | asterboy: we'll host for pot |
06:12.34 | Qwell | So... |
06:12.37 | X-Rob | why the smeg would you want to live in the US anyway? |
06:12.38 | asterboy | LOL |
06:12.41 | asterboy | host for pot |
06:13.00 | Qwell | anybody happen to know how to connect a 20 pin 286 harddrive up to a newish computer? :D |
06:13.01 | MikeJ[Laptop] | SwK, no, first time I had to touch it was today, when I figured out how badly it's set up.. |
06:13.06 | austinnichols101 | try our weedserver plan |
06:13.06 | SwK | hah |
06:13.15 | MikeJ[Laptop] | small office with a mail server in house... |
06:13.24 | asterboy | like an ESDI drive? |
06:13.31 | Qwell | asterboy: no bloody clue |
06:13.41 | brookshire | qwell: minix? |
06:13.46 | Qwell | brookshire: ...no clue |
06:13.53 | Qwell | probably some form of DOS |
06:13.53 | MikeJ[Laptop] | their mail relays through a spam filter company, to interland, then the clients pop it to their pc's... then it gets saved on the server from the clients |
06:14.07 | Qwell | ~1982 |
06:14.09 | asterboy | bet that thing makes a good bookend or doorstop |
06:14.14 | MikeJ[Laptop] | ~1492 |
06:14.15 | Qwell | I can't date it, really |
06:14.31 | Qwell | let's put it this way... |
06:14.52 | X-Rob | Qwell, 20 pin? That's MFM/RLL? |
06:14.52 | Qwell | there are posts on google groups, dating back to 1990, about how people were successful in putting a 20mb hd into the machine |
06:14.55 | asterboy | I'd think you could get some sort of ISA Card to hook that up. |
06:15.07 | Qwell | X-Rob: If I knew, I wouldn't be asking. :) |
06:15.08 | asterboy | Run Length Limited |
06:15.19 | X-Rob | Qwell, does it have edge connectors, or pins on the hdd? |
06:15.21 | X-Rob | (take a photo of it) |
06:15.25 | Qwell | edge connectors? |
06:15.30 | austinnichols101 | qwell: do you have the old edge connector cable? |
06:15.32 | Qwell | like old floppies has? |
06:15.42 | X-Rob | Qwell, yeah, the connector slides over the board |
06:15.44 | Qwell | it has two types...it's using the pines |
06:15.45 | Qwell | pins |
06:15.46 | X-Rob | kinda like a PCI slot |
06:15.52 | brookshire | qwell: it would probably be easier to buy an old irix box.. and install it |
06:15.52 | brookshire | lol |
06:16.00 | Qwell | brookshire: I want the data off the drive :p |
06:16.01 | brookshire | use it as a fileserver |
06:16.14 | oej | brookshire: You are up early :-) |
06:16.18 | X-Rob | Ooh oej |
06:16.23 | brookshire | oej: heh.. |
06:16.28 | Qwell | X-Rob: it does have that type though, yes |
06:16.29 | oej | X-Rob: Morning |
06:16.52 | Qwell | about 15 "pins", then a break, and 5 more "pins", on a board extending out |
06:16.55 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
06:17.24 | X-Rob | Qwell, that's MFM or RLL |
06:17.32 | X-Rob | you'll need to ebay for that |
06:17.38 | X-Rob | it'll be an ISA card |
06:17.41 | Qwell | They have pci cards for that? |
06:17.46 | X-Rob | nope |
06:17.46 | austinnichols101 | qwell: you're going to probably need the original card it was formatted with |
06:17.49 | Qwell | lame...I have the isa card in the old box |
06:17.58 | Qwell | but...nowhere to put it |
06:17.58 | asterboy | Qwell, here ya go: http://cgi.ebay.com/Everex-EV332-16Bit-ISA-Floppy-Hard-MFM-Drive-Controller_W0QQitemZ8792366847QQcategoryZ1247QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
06:18.00 | Qwell | it's HUGE |
06:18.16 | asterboy | 9.99$ |
06:18.21 | asterboy | Place your bid! |
06:18.28 | asterboy | AND |
06:18.31 | X-Rob | ooh yeah, the formatting was tied to the controller card |
06:18.33 | asterboy | from NEW ZEALAND |
06:18.34 | Qwell | asterboy: I already have one :D |
06:18.38 | asterboy | lol |
06:18.47 | Qwell | The one it was formatted with, I'd guess |
06:18.51 | austinnichols101 | qwell: now all you need is an old 286 |
06:18.57 | X-Rob | Qwell, so you have the card, and the drive, so what are you missing? |
06:18.59 | Qwell | austinnichols101: I have an old 286 :P |
06:19.02 | X-Rob | an ISA motherboard? |
06:19.08 | Qwell | X-Rob: no, it's a full machine |
06:19.25 | X-Rob | Qwell, I'm puzzled what your problem is then |
06:19.35 | asterboy | any ISA should do it, no? |
06:19.37 | Qwell | X-Rob: I don't think it powers on, heh |
06:20.03 | X-Rob | Qwell, ahha. Well, grab an old IBM 300GL on ebay, they've got ISA slots |
06:20.09 | X-Rob | usually about $10 |
06:20.10 | asterboy | catch |
06:20.32 | Qwell | I need to get a NIC for it, or something... |
06:20.38 | Qwell | it's got like a 240 baud modem :D |
06:20.40 | austinnichols101 | and some thin ethernet cable |
06:20.56 | Qwell | bigmouth...oh yeah |
06:20.57 | asterboy | transfer over 5.25" Disks |
06:20.59 | X-Rob | Qwell, the IBM's are Pii's and come with an onboard NIC. |
06:21.12 | X-Rob | boot knoppix, it understands MFM and RLL |
06:21.16 | *** join/#asterisk Tili (i=Tili@61.140.191.181) |
06:21.20 | asterboy | seriously? |
06:21.22 | Qwell | X-Rob: My wife would probably kill me. :) |
06:21.31 | X-Rob | Qwell, for buying a $10 computer? |
06:21.34 | Qwell | though, I will check it out |
06:21.36 | Qwell | X-Rob: yes, heh |
06:21.47 | X-Rob | put your foot down |
06:21.51 | Qwell | Question is, will the ibm fit an extended length card? |
06:21.57 | X-Rob | say 'I want this computer! And I'm not getting any sex until you forgive me for getting it!' |
06:22.10 | asterboy | if knoppix can understand MFM & RLL, I'd be so impressed. |
06:22.21 | Qwell | asterboy: linux kernel can...why wouldn't knoppix? |
06:22.22 | X-Rob | asterboy, uh, yes, it does. |
06:22.35 | asterboy | never knew that. |
06:22.44 | asterboy | should pay attention to the menuconfig options. |
06:22.55 | Qwell | So, knoppix would pick up the isa controller card, and then the drive? |
06:22.59 | Qwell | s/would/should/ |
06:23.09 | X-Rob | Qwell, yup |
06:23.12 | Qwell | neat |
06:23.15 | asterboy | very |
06:23.18 | X-Rob | the isa card always lives at c800 |
06:23.42 | X-Rob | so it looks there, goes 'ooh, I have a controller card' and hooks into it. It's insanely slow, as it uses like int13 to access it |
06:23.51 | Qwell | heh |
06:23.52 | austinnichols101 | debug: g=c800 |
06:24.06 | Qwell | I knew I asked in the right place... |
06:24.07 | X-Rob | austinnichols101, hah. how to format an old hdd 8) |
06:24.12 | Qwell | bunch of old telephony guys in here :p |
06:24.20 | austinnichols101 | don't forget to pick the right interleave |
06:24.25 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
06:24.27 | X-Rob | I'm not old, OR telephony |
06:24.27 | X-Rob | ok |
06:24.28 | X-Rob | I'm old. |
06:24.34 | X-Rob | but I'm old and linux-y 8) |
06:24.43 | asterboy | *nixy |
06:24.47 | X-Rob | true |
06:24.51 | X-Rob | SunOS 4.1 ho! |
06:25.03 | asterboy | ewww |
06:25.09 | FuriousGeorge | i used beos once |
06:25.13 | asterboy | lol |
06:25.14 | austinnichols101 | out of curiosity, what age is old now? |
06:25.26 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
06:25.27 | asterboy | grey pubic hair |
06:25.32 | MGSsancho | ew |
06:25.33 | Qwell | austinnichols101: If you can remember terms such as "MFM and RLL"... |
06:25.35 | FuriousGeorge | yuck |
06:25.37 | austinnichols101 | yeah |
06:25.44 | kmilitzer | Morning everyone ... |
06:25.47 | Qwell | There's a good chance you're old. ;) |
06:25.53 | austinnichols101 | 40 next week |
06:25.59 | X-Rob | I can't remember what MFM stands for, but RLL was 'Run Length Limited' |
06:26.07 | Qwell | 40's not so old |
06:26.15 | austinnichols101 | I'm going to have a 'halftime' party |
06:26.28 | freat | hello |
06:26.29 | Qwell | ~mfm |
06:26.30 | jbot | A graphical frontend for mtools. URL: http://www.core-coutainville.org/mfm/ |
06:26.41 | asterboy | Modified Frequency modulation |
06:26.49 | X-Rob | that's it |
06:26.51 | X-Rob | go wikipedia |
06:26.53 | SwK | ST506 y0 |
06:27.07 | SwK | Run Length Limited ++ |
06:27.23 | freat | anyone recommend a solution: asterisk server is remote. got Polycom phones on site (obviously) that I would like to be able to connect to an ATA or the like for 911 |
06:27.41 | SwK | freat: spa-3000 |
06:27.59 | freat | SwK: it will handle the phone's SIP connections? |
06:28.10 | X-Rob | freat, SPA3000 is your only option, really. |
06:28.23 | freat | X-Rob: thanks I will read up on it |
06:28.24 | X-Rob | if sipura didn't have one, someone else would have made one by now |
06:28.25 | *** join/#asterisk BenderNZ (i=bender@nz1.recoil.net.nz) |
06:28.42 | SwK | it wont play sip proxy but if it cant contact the sip proxy (asterisk) it connects the FXS port on it to the FXO port for a life line |
06:28.54 | *** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net) |
06:29.05 | asterboy | st506...I remember those...great drive |
06:29.24 | freat | SwK: hmm... so they would have to pick up the phone that's connected to it? Sure would be nice to config the polycoms to hit it somehow |
06:29.46 | BenderNZ | hi - I've got asterisk setup and incoming calls are coming in via SIP fine and my phones are ringing, however outgoing calls ring once then in the asterisk log I see Failed to authenticate on INVITE |
06:30.02 | freat | Polycoms allow for a separate server config for emergency dialing |
06:30.10 | BenderNZ | I can't see why they aren't authenticated though because they're registering and they ring when I ring them |
06:30.18 | Qwell | So, does 1982 sound about right for that hardware? |
06:30.53 | wasim | i got a vic-20 in 1982 |
06:31.08 | wasim | my first real honest to goodness computer |
06:31.54 | asterboy | vic was good, but c64 was a god send |
06:32.02 | X-Rob | Hmmmm. 1982 sounds early, but possible |
06:32.04 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
06:32.13 | X-Rob | I'd think more like 83-84 |
06:32.21 | X-Rob | what's the model number of the HDD? |
06:32.33 | Qwell | X-Rob: no clue...haven't bothered taking it out |
06:32.46 | Qwell | but it's from a Zenith z-152 |
06:33.19 | X-Rob | oooh |
06:33.23 | X-Rob | I used to sell them! |
06:33.26 | Qwell | hah |
06:33.38 | X-Rob | I got better! |
06:33.43 | FuriousGeorge | anyone wanna guesstimate *'s market share? |
06:33.53 | X-Rob | FuriousGeorge, - 100% |
06:33.55 | Qwell | So, yeah...a 20mb was a major upgrade, as was 640k ram |
06:34.00 | X-Rob | no, um 672% |
06:34.02 | austinnichols101 | 82 = vic 20 |
06:34.10 | FuriousGeorge | X-Rob: stop liein' |
06:34.24 | FuriousGeorge | do you think it reaches 1% |
06:34.25 | FuriousGeorge | ? |
06:34.28 | austinnichols101 | 84 = mac 128 |
06:34.31 | X-Rob | FuriousGeorge, of what? |
06:34.50 | FuriousGeorge | of total pbx installed in the universe |
06:34.51 | asterboy | when was the apple iie? |
06:34.56 | X-Rob | FuriousGeorge, shit no. |
06:35.23 | X-Rob | that's bazillions of 20 year old pabx's doing their stuff |
06:35.25 | FuriousGeorge | true |
06:35.32 | austinnichols101 | we made a shitload back then selling ast research six-pack plus cards |
06:35.38 | asterboy | 84 =apple iie |
06:35.41 | FuriousGeorge | i got one sitting on top of an * server :) |
06:35.47 | austinnichols101 | and tallgrass technologies disk/tape |
06:35.56 | X-Rob | I'd say it would be 5-10% of SIP PABX's though. |
06:36.16 | FuriousGeorge | whats the A for |
06:36.20 | FuriousGeorge | ~pabx |
06:36.21 | jbot | i guess pabx is Private Automatic Branch eXchange |
06:36.21 | asterboy | I just upgraded a Nortel Vantage 12 |
06:36.26 | X-Rob | Automatic |
06:36.36 | asterboy | Anyone working on Vantage? |
06:37.19 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
06:37.26 | X-Gen | ho hum |
06:38.07 | asterboy | Remember the Trash 80s? |
06:38.14 | SuperLag | Vic20++ |
06:38.22 | asterboy | http://www.old-computers.com/museum/computer.asp?c=409&st=1 |
06:38.25 | FuriousGeorge | me had an 8088 tandy |
06:38.34 | asterboy | tandies were dandy |
06:38.41 | FuriousGeorge | i think it was the tandy 1000 ir sinmething |
06:38.55 | FuriousGeorge | hell yeah, it had a 3.5" floppy |
06:39.23 | asterboy | I had an Adam, but took it back along with my garbage Commodore 128 |
06:39.57 | FuriousGeorge | i remember looking at that floppy in 1986 and thinking "i wonder what we'll be using in 20 years. then 20 years later i bought a mb and the sw raid driver for winxp is on a floppy |
06:40.23 | asterboy | that's funny |
06:40.44 | austinnichols101 | get your hayes 1200 longcard and fire up your bbs: http://software.bbsdocumentary.com/ |
06:40.53 | Qwell | 1200, pfft |
06:41.20 | Supaplex | I still have 4 8port 9600 bps multiport serial cards. =-) |
06:41.25 | asterboy | There must be some die hard BBSs running still. |
06:41.27 | Supaplex | isa even |
06:41.34 | X-Rob | FuriousGeorge, pfft. The _REAL_ TRS80's were the old Z80's. Astoundingly good machines. |
06:41.48 | FuriousGeorge | come to think of it, if windows xp was able to pull a driver off a pendrive for install, there would be absolutely no reason to use'em anymore (finally) |
06:41.50 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
06:41.52 | asterboy | coleco! |
06:41.56 | FuriousGeorge | X-Rob: i was too young for those |
06:42.13 | Supaplex | gameboy uses z80 |
06:42.46 | FuriousGeorge | at the time the only thing i knew about BASIC was that it was that annoying screen that came up when the game cartridge wasnt loaded right in my commodore 64 |
06:43.04 | Supaplex | 10 goto 10 |
06:43.06 | Supaplex | run |
06:43.13 | FuriousGeorge | and i couldnt play jungle hunt |
06:43.14 | asterboy | I sure got to know basic in a hurry in order to hack games. |
06:43.43 | asterboy | hex editor for the 1541 |
06:43.43 | FuriousGeorge | on what the old commodore? |
06:44.02 | asterboy | yep |
06:44.29 | FuriousGeorge | i knew some kid in spain, lived in the middle of nowhere, had some old comuter that ran on data cassettes. as in audio cassettes with data on them |
06:44.42 | asterboy | so that was say 20 year ago...can you imagine in the next 20? |
06:45.00 | austinnichols101 | I had the cassette drive for my vic 20 |
06:45.01 | FuriousGeorge | the floppy's should hold 1.44 megs by then |
06:45.29 | asterboy | The Timex Sinclair (Z81) had a cassette |
06:45.32 | wasim | a nice white cassette tape recorder |
06:45.40 | asterboy | lol, nice white |
06:45.46 | *** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net) |
06:45.48 | wasim | not the silly beige of the c64 |
06:45.59 | asterboy | that would turn orange if left in sunlight |
06:46.14 | asterboy | dig dug |
06:46.22 | asterboy | M.U.L.E. |
06:46.25 | FuriousGeorge | dig-dug was great |
06:46.38 | asterboy | Indiana Jones |
06:46.40 | Qwell | X-Rob: If you can remember the z152...how about the "centaur II"? |
06:46.42 | asterboy | Bruce Lee |
06:46.56 | X-Rob | Qwell, nope, that doesn't ring a bell. |
06:46.57 | FuriousGeorge | i had indiana jones too but i dont remember playing it |
06:47.06 | X-Rob | isn't centaur a VIA cpu? |
06:47.07 | asterboy | it had a snake pit |
06:47.16 | austinnichols101 | isn't that the one where you would swing with the whip? |
06:47.19 | Qwell | dunno |
06:47.21 | asterboy | Moon Lander |
06:47.28 | asterboy | err...Moon Rover |
06:47.58 | asterboy | Qbert |
06:49.34 | Supaplex | ok, the $brain is drained. time for bed(); nite. |
06:49.42 | asterboy | hear ya there. |
06:50.01 | FuriousGeorge | i used to hate when jumping on a block 1 extra time turned it the wrong color |
06:50.09 | asterboy | great to reminice. |
06:50.35 | FuriousGeorge | although i do still play cupert on the nintendo emulator |
06:50.38 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
06:50.55 | FuriousGeorge | its legal cuz i had the game and the system, OK |
06:51.01 | asterboy | seriously, after remembering all that, I can't believe where I am in computers and technology today. |
06:51.06 | FuriousGeorge | as for the other 700 games that came in the 10 meg file |
06:51.14 | FuriousGeorge | i deleted those :) |
06:51.16 | asterboy | it really does seem like a dream |
06:51.28 | asterboy | ya I had a ton of games. |
06:51.37 | asterboy | disk upon disk |
06:51.52 | asterboy | some good porn ones too |
06:51.58 | FuriousGeorge | asterboy: you can get them all again. they're in a torrent somewhere |
06:52.03 | FuriousGeorge | i bet Qwell is hosting it |
06:52.23 | asterboy | I know, there are some amazing sites doing java emulation of the C64 chip |
06:52.36 | Qwell | 10mb? please, my zenith can't even hold that much :p |
06:52.37 | FuriousGeorge | dont get me started please |
06:52.52 | FuriousGeorge | lol |
06:53.05 | asterboy | Now look at battlefield 2 and the next gen of games. |
06:53.18 | asterboy | online gaming is a culture all on its own. |
06:53.30 | tainted- | that's deep |
06:53.53 | asterboy | that it is. |
06:54.07 | FuriousGeorge | brb |
06:54.11 | asterboy | I'll fade to black on that note. |
06:54.32 | asterboy | Building an * box in the morrow |
06:54.58 | tainted- | men amongst men |
06:54.58 | asterboy | thanks for the step back into computer history. |
06:55.04 | asterboy | yes |
06:55.08 | tainted- | :D |
06:55.25 | asterboy | next ime lets go back to Charles Babbage...the supposed father of computers. |
06:56.20 | asterboy | well..you could say the abbacus started the road. |
06:56.43 | asterboy | night guys. |
06:56.47 | tainted- | night |
07:03.27 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
07:04.26 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-48.claranet.co.uk) |
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07:20.46 | *** join/#asterisk tetsuzan (n=raizen@200.180.124.12) |
07:22.27 | tetsuzan | anyone knows if zaptel already works with freebsd 6? |
07:22.42 | zoa | dont think so |
07:22.53 | tetsuzan | time ago, i had test |
07:23.03 | tetsuzan | and, it fails |
07:24.41 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
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07:28.16 | *** part/#asterisk mogorman (n=mogorman@68.62.237.103) |
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07:35.03 | *** join/#asterisk SHad|Work (n=kvirc@popust.net) |
07:35.14 | SHad|Work | hello |
07:35.40 | SHad|Work | I'm a bit puzzled with how asterisk and sip phones use the codec settings |
07:36.14 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
07:36.55 | SHad|Work | is it normal for the phones to use just the first codec from the allow list? asterisk then always tries to transcode the calls instead of requesting the same codec on both phones |
07:41.19 | *** join/#asterisk plasko (n=plasko@triana.kmpanilla.com) |
07:41.25 | tetsuzan | you have to add the codecs as a priority list |
07:41.33 | plasko | wow its busier in here than I expected. |
07:41.42 | tetsuzan | if your phone supports the first codec, the codec will be used. |
07:42.01 | kaldemar | SHad|Work: the allow's are in order of preference. |
07:42.33 | kaldemar | SHad|Work: e.g. if your phone supports the first codec, it will use it. |
07:43.38 | plasko | has anyone used RAGI? |
07:44.34 | kaldemar | SHad|Work: do you have canreinvite=no in your sip.conf? |
07:45.59 | *** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca) |
07:46.53 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
07:47.05 | Shaun2222 | are their any advantages to using sccp? |
07:47.12 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
07:52.03 | Shaun2222 | ok.. |
07:52.47 | Shaun2222 | anyway to make the phone show parked calls, i've used a few pbx's where it shows calls parked on the phone, so far the asterisk config just tells you where it was parked and then you have to remember it i guess... |
07:57.10 | SHad|Work | kaldemar: no, I have canreinvite=yes |
07:57.42 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.190) |
07:57.46 | SHad|Work | I thought asterisk tried to negotiate a native call between the phones |
07:57.56 | Zhadnost | it depends on the config. |
07:58.07 | Zhadnost | depends on the nat= and the canreinvite= |
07:58.09 | SHad|Work | what part of the config? |
07:58.19 | SHad|Work | nat=yes, canreinvite=yes |
07:58.31 | SHad|Work | that's what I've got |
07:58.31 | kaldemar | i think it's supposed to if you have canreinvite=yes and the phones are on the same lan. |
07:58.47 | Zhadnost | then (from vague memory) asterisk assumed the phones are behinfd a NAT and can't directly talk to each other. |
07:59.07 | Zhadnost | (I'm not sure, but it's all on voip-info). |
07:59.11 | SHad|Work | but g729 works in passhtrough |
07:59.17 | SHad|Work | I don't have the g729 licence |
07:59.29 | Zhadnost | you don't need to if it's on passthough |
07:59.35 | Zhadnost | even if the call is being routed through asterisk |
07:59.50 | x86 | morning :) |
08:00.01 | Zhadnost | the SVN branch supports T.38 in passthrough but that doesn't mean asterisk has any idea aregarding the traffic. |
08:00.10 | SHad|Work | so is there any possibility of knowing if the calls are routed through asterisk? |
08:00.45 | Zhadnost | I guess it'd appear in sip show channels |
08:01.03 | SHad|Work | so the nat setting might be the culprit |
08:01.07 | Zhadnost | also (from memory) asterisk wont pass over the call if you have other items configured, like interactive calling or recording). |
08:01.45 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
08:01.50 | Shaun2222 | is ztdummy needed with 2.6? |
08:02.15 | Telamon | shaun222: If you don't have a TDM or TE card, then yes, I believe so. |
08:02.34 | Zhadnost | if you want to use Meetme rooms, MoH etc. |
08:02.44 | Shaun2222 | k |
08:02.45 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
08:03.00 | Telamon | ztdummy is used to provide a timing source. If you have a real Digium PRI card, it uses the timing from that instead. |
08:03.20 | SHad|Work | so in regard to scallability if I use MoH the server network infrastructure should be able to handle all the calls bandwidth |
08:03.25 | SHad|Work | never thought about that |
08:03.27 | SHad|Work | thank you |
08:03.45 | SHad|Work | I'll try the nat setting as soon as get the server up again |
08:04.12 | Telamon | Anyone know why my Asterisk would be crashing when running certain macros? Is there a limit to the number of macros you can call? IE, macro a calls macro b two or three times. |
08:05.50 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
08:06.09 | Zhadnost | weird thing happened on Sunday, I put a TDM400 card in a machine, configured it and ran ztcfg, ran asterisk and everything was fine. A couple of hours later asterisk had stopped running and wouldn't start (card hadn't been configured) zttool confirmed that the card hadn't been configured and then after runnign ztcfg again everything was fine. |
08:06.16 | Zhadnost | Did I miss a step in the setup? |
08:07.32 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
08:09.20 | Telamon | Zhadnost: Did you reboot or reload any modules in between the first ztcfg run and the crash? |
08:09.48 | Telamon | reload = remove or add, not necessarily just remove and add back in |
08:10.09 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
08:10.10 | Zhadnost | I don't remember doing it, I may have done an asterisk reload or similar. |
08:10.40 | cced | Ariel_ :hi |
08:10.52 | Zhadnost | It hasn't happened since |
08:11.24 | Telamon | Hmm, if you unload the module (which I don't think you can do with Asterisk running, as the device would be in use) or possibly if you load another ZT module it might cause problems. Or it might have just been gremlins. :) |
08:11.46 | *** join/#asterisk sercz (n=serz@i3ED6F0DD.versanet.de) |
08:16.47 | DoktorGreg | I remember MFM and RLL though i dont remember what they mean |
08:17.27 | Zhadnost | I'm betting on cosmic rays |
08:18.26 | DoktorGreg | gremlins are real! |
08:18.52 | DoktorGreg | you can have zero stress on a cable |
08:19.03 | DoktorGreg | but if you dont have it battened down, it will get pulled out |
08:19.13 | DoktorGreg | when there is no possible way for it to get pulled out |
08:19.29 | DoktorGreg | I have experienced this effect a lot |
08:19.41 | Zhadnost | that sounds like an office to me |
08:19.49 | DoktorGreg | now everything gets a zip tie |
08:20.09 | DoktorGreg | zip tie's are gremlin bane |
08:20.56 | DoktorGreg | holy shite, boot camp has made me insane |
08:21.11 | DoktorGreg | i just read mac vs pc blogs for 4 hours |
08:21.17 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
08:21.41 | Zhadnost | eww |
08:22.05 | DoktorGreg | my first computer after my atary 800 was a mac 512 |
08:22.34 | DoktorGreg | ive been bent ever since then |
08:22.51 | Telamon | MFM and RLL were old disk drive standards, before ATA and SCSI came around, if memory serves me correctly. |
08:23.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
08:23.15 | Zhadnost | the eww wasn't in reference to macs, more the reading articles comparing them for 4 hours. |
08:23.37 | DoktorGreg | lol, like i said Boot camp made me insane |
08:23.42 | Zhadnost | did it go along the lines of, my dicks bigger than yours, yeah but mine has a go faster stripe on it? (etc. etc.). |
08:24.03 | wasim | 11:26 < asterboy> Modified Frequency modulation |
08:24.08 | wasim | and Run Length Limited |
08:24.09 | DoktorGreg | um, again, I want a mac |
08:24.16 | DoktorGreg | i really want one |
08:24.21 | Zhadnost | a mini? |
08:24.21 | DoktorGreg | but logically i cant justify it |
08:24.32 | DoktorGreg | no i want a core duo |
08:24.35 | DoktorGreg | imac |
08:24.38 | DoktorGreg | 2 ghz |
08:25.01 | DoktorGreg | but i keep specing computers twice as fast for same mony |
08:25.14 | DoktorGreg | or... |
08:25.36 | DoktorGreg | a couple of computers, same as fast... |
08:26.09 | DoktorGreg | and i cant do it for osx |
08:26.26 | DoktorGreg | because with... 15 years of legacy this direction.... |
08:26.54 | DoktorGreg | ... |
08:27.45 | DoktorGreg | mostly my observation is that, like me |
08:27.54 | DoktorGreg | mac users are generally insane |
08:28.04 | DoktorGreg | well the ones posting to blogs anyhow |
08:28.19 | *** part/#asterisk SyrusMPL (n=pascal@tahiti.mpl.rullier.net) |
08:28.30 | DoktorGreg | They make these wild eyed assertions like... |
08:28.34 | DoktorGreg | windows in unstable |
08:28.39 | DoktorGreg | ??? |
08:28.51 | DoktorGreg | windows hasnt been unstable for like a decade.... |
08:29.05 | DoktorGreg | well whenever it was that i started using nt... |
08:29.41 | DoktorGreg | but even 98 wasnt as bad as they make it out to be |
08:29.46 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
08:29.49 | Zhadnost | it could be |
08:30.18 | Telamon | Mac users have "drunk the cool-aid", so to speak. They don't just think their Macs are better, they *BELIEVE* their Macs are better. Personally, I use Linux and it works fine for me. |
08:30.21 | Zhadnost | Did anyone else notice that there was no fax server in 98 (and there was in 95). |
08:30.55 | Zhadnost | Oh, and it alxso isn't in any subsequent version. |
08:30.57 | DoktorGreg | I was using slackware for server things when 95 came out |
08:30.57 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
08:31.21 | Zhadnost | now I mostly use debian. |
08:31.24 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
08:31.39 | DoktorGreg | IIRC windows 95 had no TCP/IP stack installed by defaut |
08:31.41 | Telamon | Slackware. Now there was a real man's OS. Here are some tar files, unpack them and have fun... :) |
08:31.58 | DoktorGreg | slack is still one of the largest distros |
08:31.59 | Zhadnost | There was nothing Unixy at the time that fitted into the windows environment as well as the w95 Fax server though. |
08:32.31 | Zhadnost | Had IP/X by default, but that was usually changed during setup |
08:32.35 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
08:32.43 | cced | <PROTECTED> |
08:32.59 | Telamon | DoktorGreg: Yep, it's BSD with drivers. We used to be 100% Slackware servers at the office, but the upgrades were killing me, so I'm migrating to Gentoo. Still a lot of old Slack boxes kicking around our server room though. |
08:34.36 | DoktorGreg | I pretty much cut my teeth on slack, so now when i use just about anything else |
08:35.07 | DoktorGreg | i head straight for the cli, /etc and vi |
08:35.23 | DoktorGreg | Ive tried vi on windows |
08:35.36 | DoktorGreg | it just doesnt work the same anyhow |
08:35.45 | DoktorGreg | i keep meaning to work more with vim |
08:36.43 | DoktorGreg | again, boot camp has driven me over edge |
08:36.50 | DoktorGreg | lol |
08:36.56 | DoktorGreg | but dang it |
08:37.03 | DoktorGreg | they cost too darn much |
08:37.09 | Shaun2222 | where does the DIALTEMPLATE go, does it just go in the SIPDefault or SIP<MAC>.xml? |
08:37.48 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
08:38.43 | DoktorGreg | I cant figure it out |
08:38.59 | DoktorGreg | is it having that mac 512 20 years ago??? |
08:39.15 | DoktorGreg | that machine that barely worked??? |
08:39.26 | DoktorGreg | with 400kb floppy disks? |
08:39.44 | DoktorGreg | that i had to swap to get the os to load? |
08:40.29 | DoktorGreg | and i had to copy them every couple of weeks because the floppies that the os was on would wear out??? |
08:40.55 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
08:41.31 | DoktorGreg | why on earth do i have happy memories about that??? |
08:41.55 | DoktorGreg | and that little 9" screen |
08:42.19 | DoktorGreg | then one day |
08:42.19 | Zhadnost | does insanity run in the family? |
08:42.29 | DoktorGreg | hmmmm |
08:42.37 | DoktorGreg | family is all small biz people |
08:42.42 | DoktorGreg | does that count? |
08:42.57 | Zhadnost | pretty much |
08:43.16 | DoktorGreg | well we would actually be considered mid sized now |
08:43.27 | Zhadnost | cool |
08:43.33 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
08:44.20 | DoktorGreg | asterisk has been a lot of fun |
08:44.26 | DoktorGreg | hahaha check out this email i got today |
08:44.38 | DoktorGreg | oh what that copy paste server? |
08:45.11 | Telamon | pastebin.ca |
08:45.23 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
08:45.37 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
08:46.02 | DoktorGreg | http://pastebin.ca/48992 |
08:46.11 | DoktorGreg | anyhow i was trying to order a 1000 dollar thing |
08:46.31 | DoktorGreg | and i couldnt get the order to go through their insane security checking thing |
08:46.45 | DoktorGreg | because it is a business location |
08:46.53 | DoktorGreg | and we do all mail to a po box |
08:47.00 | DoktorGreg | and everyhing is shipped to a building |
08:47.16 | Zhadnost | On the third day of speaking to sales reps, one finally agreed to fax a quote/proforma. |
08:47.47 | Zhadnost | When I called him back 5 hours later, he said. 'What? You didn't really expect to get the Fax today did you?' |
08:48.00 | DoktorGreg | from voip supply? |
08:48.23 | Zhadnost | In the end we couldn't buy the kit because the fundholding manager was going to Canada the next day and by the time he'd come back it was a new financial year. (Which I had explained to the salesman, but did no good). |
08:48.55 | Zhadnost | no-one seems to like you spending lots of money with them nowadays, it's weird. |
08:49.06 | DoktorGreg | well anyhow |
08:49.21 | DoktorGreg | i didnt pay the ship same day premium from the digium site |
08:49.24 | Zhadnost | What was the security problem? |
08:49.43 | DoktorGreg | they wanted me to type in po box 239 for both shipping and billing |
08:50.02 | Zhadnost | weird |
08:50.09 | Zhadnost | the card is registered to a PO BOX? |
08:50.20 | DoktorGreg | yah, company po box |
08:50.33 | DoktorGreg | industrial district no mail address... |
08:51.42 | Zhadnost | weird |
08:52.18 | DoktorGreg | anyhow |
08:52.37 | DoktorGreg | their site said i had to call and register extra addresses to the credit card |
08:52.45 | DoktorGreg | so i called controller |
08:52.48 | Zhadnost | understandable |
08:53.04 | DoktorGreg | he was like, huh? |
08:53.41 | DoktorGreg | we have shipping address |
08:53.48 | *** join/#asterisk mtryfoss (n=mtryfoss@80.239.93.22) |
08:53.58 | DoktorGreg | usps doesnt deliver mail there |
08:54.15 | DoktorGreg | i dont know particulars |
08:54.26 | DoktorGreg | its never been a problem before |
08:54.28 | Zhadnost | strange |
08:54.50 | Zhadnost | mind you, the postal service here will moreorless deliver to anywhere |
08:55.03 | Zhadnost | If you give a drain as a postal address they'd deliver to it. |
08:55.23 | Ahrimanes | hm here they hardly deliver to actual adresses |
08:55.45 | Zhadnost | It will be like that here in a few years when the postal service is finally privatised :-( |
08:56.02 | DoktorGreg | well the best deal in town for business is fedex |
08:56.16 | DoktorGreg | fedex lets us buy shipping at wholesale rates |
08:56.31 | DoktorGreg | we then sell shipping to our customers at the standard fedex retail rates |
08:56.54 | mtryfoss | asterisk seems to dealock when originating a call into a queue (auto callback). is this a known issue ? |
08:57.02 | *** join/#asterisk slav_jb (n=k@pirus.securax.be) |
08:57.15 | DoktorGreg | We make like a 100k a year on that plan |
08:57.26 | Zhadnost | Over here rates seem to depend on the exact size and weight of a package as to the cheapest shipping, it's almost a science to work out. |
08:57.32 | Zhadnost | cool. |
08:57.34 | cced | who is in CHINA? |
08:58.24 | DoktorGreg | oh someone in china |
08:58.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:58.48 | DoktorGreg | can you do, or know someone who can, do home decor manufacturing? |
08:59.16 | Shaun2222 | anybody know what the name of a normal dialtone is with these bellcore-* tones? |
08:59.40 | DoktorGreg | whoops, got caught selling out my fellow americans again:P |
08:59.57 | Frogzoo | I'm wondering if China allows IRC? |
09:00.29 | DoktorGreg | they get Google lite in china |
09:00.36 | Zhadnost | I must say, I don't think I''ve ever spoken to someone from China on IRC. |
09:01.04 | DoktorGreg | Ive spoken to people who claim to be from china in Second life |
09:01.09 | *** join/#asterisk kisu (n=daniel@2001:618:400:0:0:0:da26:a0d2) |
09:01.25 | DoktorGreg | but its the internet so..... |
09:01.50 | DoktorGreg | and its second life |
09:02.03 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
09:02.07 | cced | <PROTECTED> |
09:02.08 | Zhadnost | with all the free proxies out there, it's hard to work out how they can block traffic. |
09:02.13 | DoktorGreg | which seems fundamentally based in the idea of being more interesting than you actually are |
09:02.17 | DoktorGreg | the game second life |
09:02.40 | Zhadnost | cced> you appear to be |
09:03.03 | cced | i want to local language help. yes. who else? I want code dev help |
09:04.39 | cced | Zhannost: where a u? |
09:04.52 | Zhadnost | UK |
09:05.02 | *** join/#asterisk vlrk (n=vlrk@59.93.77.120) |
09:05.06 | tzafrir | he can tell you, but he'll have to kill you |
09:05.22 | vlrk | does any body have idea on sipura auto provisioning |
09:05.31 | cced | Zhannost: are you familiar with libpri? |
09:05.52 | Zhadnost | tzafrir> They gigve you software for it if you buy >200 units. |
09:06.00 | Zhadnost | cced> Fraid not. |
09:06.17 | tzafrir | cced, why do you ask? |
09:06.50 | cced | tzafrir: I want to write some document about libpri |
09:07.20 | cced | about SIP signalling <-> isdn pri signalling exchange . |
09:07.22 | tzafrir | cced, first-off, try #asterisk-dev |
09:07.36 | cced | Zhannost: thanks ~ |
09:07.48 | cced | irc channel? |
09:07.56 | tzafrir | cced, also, it is generally more useful to ask a specific question |
09:08.08 | wasim | cced: http://www.packetizer.com/rfc/rfc.cgi?num=3398 |
09:08.51 | cced | thanks |
09:12.13 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
09:15.00 | dlynes | anyone know what the diff is between a digium x100p and a digium x101p? |
09:15.40 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
09:15.42 | Zhadnost | From memory, one's a motorola modem and the other one's an Intel. |
09:15.56 | Zhadnost | The x100p was suppossed to be a superior product. |
09:16.02 | dlynes | ah |
09:16.09 | dlynes | why is the 101p a higher version number then? |
09:16.19 | Zhadnost | because it was released later. |
09:16.25 | Telamon | dlynes: It was made later, with cheaper parts. |
09:16.27 | Zhadnost | and as a replacement. |
09:16.35 | dlynes | that's lame...bring out a cheaper product later? |
09:16.42 | Zhadnost | all companies do it. |
09:16.44 | dlynes | you'd think they'd improve it |
09:16.49 | dlynes | not cheapen it |
09:17.06 | Zhadnost | an SPA2000 is a cost reduced cisco 186, and a PAP2 is a cost reduced SPA2000 |
09:17.27 | dlynes | but the spa2000 is a better product than the cisco 186, isn't it? |
09:17.36 | dlynes | and the pap2 is a better spa2000 too, right? |
09:17.44 | Zhadnost | the software is, but it is still a cheaper build. |
09:17.53 | dlynes | ah |
09:18.11 | Zhadnost | I don't think the PAP2 was any better than the 2000. |
09:18.27 | dlynes | I heard the pap2 supported faxing slightly better than the 2000 |
09:18.39 | Zhadnost | nto afaik |
09:18.44 | dlynes | ah |
09:18.52 | dlynes | what about the 2100 then? |
09:18.57 | Zhadnost | the hardware is identical. (infact later SPA 2000's had the same board). |
09:19.17 | Zhadnost | funnily enough the ATA 186 could recognise fax traffic. |
09:19.36 | dlynes | dunno...my stupid sipura 2000 can't upgrade higher than 2.0.13g |
09:19.39 | Zhadnost | doesn't |
09:20.01 | dlynes | I heard the ata186 used more bandwidth than the spa2000, though |
09:20.15 | dlynes | i don't know how it's possible though, if it's using hte same codecs |
09:20.24 | *** join/#asterisk vopi (n=kkk@202.139.197.130) |
09:20.27 | Zhadnost | I don't remember seing 723/729 on it. |
09:20.40 | Ahrimanes | if only grandstream ata's would have working callerid on B&O telephones.. |
09:20.58 | Zhadnost | either way, the PAP 2, doesn't handle compressed codecs in more than one channel at a time. |
09:21.15 | dlynes | neither does the spa2000 |
09:21.30 | dlynes | atcom can handle two g729 streams? |
09:21.37 | Zhadnost | 4 channels, all of which can simulateously use G.729 |
09:21.42 | dlynes | ah |
09:21.43 | dlynes | cool |
09:21.49 | dlynes | how much are they? |
09:22.08 | Zhadnost | Only problem is, the firmware is buggy (if it gets a NOTIFYU command from asterisk, it falls over). |
09:22.13 | dlynes | oh |
09:22.20 | Zhadnost | quite expensive |
09:22.36 | Zhadnost | (basically means you don't configure a mailbox in sip.conf). |
09:22.39 | dlynes | so, a digium tdm400 cheaper? |
09:22.53 | dlynes | s/cheaper/is cheaper |
09:23.46 | Zhadnost | $119.99 |
09:23.55 | Zhadnost | proba bly down to $100 if bought from the factory |
09:24.05 | dlynes | ' $100/port? |
09:24.11 | dlynes | ' or $100 for all four? |
09:24.14 | Zhadnost | no, it's an ATA, that's all 4 ports. |
09:24.21 | Zhadnost | 4 X FXS ports |
09:24.24 | dlynes | I thought you said it was expensive? |
09:24.28 | dlynes | That's dirt cheap |
09:24.36 | Zhadnost | that is when you're talking about grandstreams |
09:25.01 | dlynes | It's about $100 for a sipura unit that only has two ports |
09:25.11 | dlynes | Well...$110 |
09:25.16 | dlynes | $CDN |
09:25.21 | Zhadnost | if you buy from the factory you will probably need to order 20 at a time, and the factory is in Shenzhen |
09:25.37 | dlynes | ah...that's where I've seen the name before |
09:25.44 | Ahrimanes | Zhadnost: do atcom have an english product page for that ata? |
09:26.00 | dlynes | They're always sending me catalogues from China and Taiwan |
09:26.24 | Zhadnost | Ahrimanes> http://www.atcom.cn/En_products_AG468.html |
09:26.38 | Ahrimanes | Zhadnost: thanks |
09:26.58 | Zhadnost | It's not based on the PA1688 (like the rest of their products) It's a Myson Century CS3220 |
09:27.12 | *** join/#asterisk sercz (n=serz@i3ED6F067.versanet.de) |
09:28.01 | vlrk | Any idea on "sipura spa 841 auto provisioning" ? |
09:28.03 | dlynes | http://www.atcom.com.cn/ is the Chinese site; http://www.atcom.cn/ is the English site |
09:28.18 | vopi | ahhh I have thats model |
09:28.23 | vopi | AG468 |
09:28.35 | vopi | but it is sleeping |
09:28.41 | dlynes | PA1688...isn't that based on something from Advantage Century Telecom? |
09:29.01 | Zhadnost | All I can tell you regarding the T.38 support is that if you plug a fax machine into it, it will detect the fax signal and request T.38. (Can't really test any further than that). |
09:29.05 | Ahrimanes | Zhadnost: hm.. the 4 ports.. can they be configured for 1 number or only as seperate numbers? |
09:29.30 | dlynes | Ahrimanes: That would all depend on your asterisk configuration |
09:29.44 | Zhadnost | Ahrimanes> you get to comfigure 2 servers, and set usernames and passwords for each port (second server as a backup server). |
09:30.41 | Zhadnost | So you can do it by configuring asterisk to let all 4 ports use the same account or set up accounts that are equivalent in asterisk, there is nothing inherent in the ATA. |
09:31.08 | Zhadnost | afaik you can't route between the ports with a dialplan like you can with sipura stuff |
09:31.11 | *** join/#asterisk julesvd (n=julesvd@pdpc/supporter/student/julesvd) |
09:31.11 | Ahrimanes | Zhadnost: ok thanks.. hm i need a 3+ ata that just has the same account for all ports.. damnit |
09:31.27 | julesvd | hi all ! |
09:31.45 | Ahrimanes | oh well, i should get to the office.. ttyl |
09:32.54 | *** join/#asterisk Tili (i=Tili@218.19.160.14) |
09:34.02 | *** join/#asterisk yuvan_k (n=scowser8@219.95.78.33) |
09:34.48 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
09:35.27 | backblue | hi, anyone with x100p in europe, working with CLIP ( callerid ) ? |
09:38.31 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
09:39.01 | *** part/#asterisk BenderNZ (i=bender@nz1.recoil.net.nz) |
09:39.43 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
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09:44.32 | *** part/#asterisk yuvan_k (n=scowser8@219.95.78.33) |
09:45.27 | zoa | i never got the callerid to work on those crappy x100ps |
09:48.38 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
09:49.48 | backblue | exacly my problem too. |
09:54.14 | kaldemar | would anyone know where the "starting context" of a call is defined, and whether it can be changed from the dialplan? |
09:54.31 | kaldemar | by starting context i mean the context where for example the hangup extension is searched in when either the caller or the callee hangs up. |
09:56.27 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-10-31.w86-207.abo.wanadoo.fr) |
10:04.54 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
10:05.00 | Mystiq | kaldemar: for an incoming call ? check zapata.conf, iax.conf or sip.conf |
10:06.26 | kaldemar | Mystiq: no no, they're all incoming in asterisk's sense. |
10:07.23 | kaldemar | i know all the context=blablah rows in conf files, but when you change the current context in the dialplan with app Goto, it changes the context that is seen as the start context for the call. |
10:07.44 | kaldemar | and it doesn't find the hangup context. |
10:08.52 | kaldemar | i've found an ugly solution for that already, but... it's ugly. |
10:10.20 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
10:13.08 | Mystiq | you need to have a hangup extension in every context ofcourse |
10:14.10 | RoyK | Mystiq: i beleive he means if starting off in [a], goto(b,s,1) then [a]'s hangup extension is run when the call is hung up |
10:15.11 | backblue | zoa: which country are you in? |
10:15.28 | dlynes | RoyK: but it's currently looking for a hangup extension in [b], not [a] |
10:15.41 | dlynes | RoyK: because he entered the [b] context which is included in the [a] context |
10:15.57 | dlynes | RoyK: and then issued a dial command from the [b] context |
10:16.12 | *** join/#asterisk Vahram (n=nospam@83.139.4.112) |
10:16.22 | dlynes | right, kaldemar? |
10:16.51 | kaldemar | not exactly, otherwise, but b isn't included in a. |
10:16.58 | kaldemar | i'm using a goto to b from a. |
10:17.10 | dlynes | Yeah, so it switches contexts |
10:17.15 | *** join/#asterisk UnderMine (n=paddy@host81-134-109-39.in-addr.btopenworld.com) |
10:17.23 | UnderMine | lo |
10:17.23 | kaldemar | i'm trying to avoid putting the hangup context in every context. |
10:17.25 | dlynes | it's included in a, just not using the include statement |
10:17.36 | kaldemar | dlynes: yes. |
10:17.38 | Mystiq | RoyK: ehm, if you goto to b, then the hangup extension in b should be run |
10:17.40 | dlynes | so it's not actually part of hte a context, but you get my point |
10:17.50 | *** join/#asterisk h3x0r (i=hex@ip68-96-175-172.lv.lv.cox.net) |
10:18.20 | dlynes | kaldemar: this is where objects would come in handy |
10:18.34 | dlynes | make every context inherit the hangupable context :) |
10:18.40 | kaldemar | Mystiq: it is, but as i said i don't want to have hangup extension in every context because i have a hierarchical structure and it would be so pretty if i only could have it in one context. :) |
10:18.45 | RoyK | kaldemar: and then which hangup extension is run? |
10:19.22 | Mystiq | kaldemar: no other way.. you would need a hangup extension in every context, could be a Macro, but you need a hangup extension i'm afraid |
10:19.22 | kaldemar | RoyK: exactly. at the moment one hangup extension is enough for every call class, but in the future... |
10:19.49 | kaldemar | that's why i'd like to see and option for that in the dial app. |
10:20.12 | dlynes | kaldemar: You could always write it, and then submit it to digium for inclusion in the next version of asterisk |
10:20.17 | kaldemar | that you could trigger a macro when the call gets hung up, no matter if it's the caller or the callee that hangs up. |
10:20.21 | Mystiq | not related to dial i think, would be pbx.c |
10:20.29 | syle | anyone have the patch for ibm's pthread error that happens with asterisk 1.2.x for opensouce g729? |
10:20.41 | Vahram | syle, yep |
10:20.51 | syle | can you please send to me |
10:20.58 | Vahram | shure |
10:21.01 | syle | ty |
10:21.03 | kaldemar | Mystiq: it is in pbx.c, but it would be a nice feature in app_dial |
10:21.19 | kaldemar | dlynes: i could write it if i could write c. |
10:21.47 | kaldemar | maybe i'll just have to quit whining and learn me some. ;) |
10:29.21 | grem_lin | Hi there, could someone possibly give me some help/advice on my dialplan http://pastebin.ca/49001 . My problems are that when you enter an invalid extension, it seems to be going to _X. again - I would use 's' however when I dial the inbound number I get no reply and it simply "times out". Also, is there a way to differenciate from no reply and an invalid 2xx extension, which are defined in my [Outgoing] context? |
10:32.42 | backblue | callerid with x100p? anyone? |
10:34.23 | *** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) |
10:35.27 | wmandra | morning all |
10:35.57 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
10:37.44 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
10:39.16 | grem_lin | Does anyone have any idea why the 's' extension might not be working as one would expect it to? Any help would be greatly appreciated... |
10:41.39 | sternn | grem_lin: I'm not positive, but you might check your capitalization for the "i" section Playback vs. playback, etc. |
10:43.13 | grem_lin | Surely I'd see that on the console though? I beleive that it's not actually reaching the "i" section at all, and the _X. is picking up on anything at all, which is why I'm thinking I need to try and get the "s" extension to work to take the initial call |
10:46.10 | *** join/#asterisk cj-rm (n=cjrm@81-86-30-78.dsl.pipex.com) |
10:46.17 | cj-rm | Hey people... |
10:46.57 | cj-rm | For some reason I'm getting problems with Asterisk detecting answered calls when making an outgoing call on an FXO line via a TDM400. |
10:46.59 | backblue | UnderMine: easy cake. :P |
10:47.02 | cj-rm | Anyone any ideas??? |
10:49.04 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
10:50.40 | andrebarbosa | anyone can help me with Huawei softswitch integration? (SIP) |
10:50.47 | cj-rm | hmmm.... Ok, I'm in the UK (on BT) and I've just set callprogress=no and it works, but asterisk doesn't wait for the call to be answered... Any ideas on how to fix this, callprogress=yes halts execution of the dial plan after the Dial() but doesn't detect the call being answered... |
10:50.52 | cj-rm | is there a UK setting for this?? |
10:51.11 | andrebarbosa | http://forums.digium.com/viewtopic.php?t=5751&highlight=huawei |
10:51.32 | andrebarbosa | if anyone have similar problems let me now please :) |
10:51.39 | andrebarbosa | had* |
10:54.26 | cj-rm | Does anyone know if there are any unit testing frameworks for asterisk?? |
10:57.40 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
11:01.45 | ManxPower | cj-rm, define unit testing |
11:02.10 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
11:02.32 | UnderMine | backblue: not when the C2 card didn;'t want to talk |
11:02.46 | ManxPower | The docs for WaitExten suck |
11:04.07 | *** join/#asterisk shiznatix (n=shiznati@213-35-236-128-dsl.end.estpak.ee) |
11:04.36 | shiznatix | can anyone give me a list of all the characters that can be used in the dialplan that asterisk will recognize? |
11:05.01 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
11:05.22 | rkr245 | hello |
11:05.25 | rkr245 | and hi |
11:07.48 | backblue | shiznatix: read the rfc, it depends on the protocol |
11:09.32 | shiznatix | backblue, what is the rfc |
11:11.18 | backblue | shiznatix: do you work in this area, or just for fun? |
11:13.08 | Mystiq | anyone installed an asterisk (using zap) in ukraine ? apparently you need some kind of certification for that ? |
11:13.25 | shiznatix | backblue, just for fun really |
11:15.08 | ManxPower | shaun222, All printable ASCII characters |
11:15.16 | ManxPower | ..er.. |
11:15.23 | ManxPower | shiznatix, All printable ASCII characters |
11:15.42 | shiznatix | ManxPower, thanks |
11:15.57 | ManxPower | If you give more details about specifically WHAT you are asking about. are you asking about extensions, pattern matches, Dial options? |
11:18.02 | rkr245 | hi can any body tell the process of adding friends in sip.conf and extensions .conf with ip address |
11:19.09 | shiznatix | ManxPower, patterns for the number that was dialed like: _95XXXXXXX. then I also need for the extension do dial like: {$EXTEN:1}. |
11:19.37 | shiznatix | ManxPower, basically what chars will be used by asterisk in those 2 situations so I can make sure nothing else is put in |
11:19.40 | ManxPower | shiznatix, You mean dial=> ${EXTEN:},1,Dial(.... |
11:19.48 | ManxPower | ..er |
11:19.55 | ManxPower | shiznatix, You mean exten => ${EXTEN:},1,Dial(.... |
11:20.10 | shiznatix | ManxPower, yes |
11:20.25 | ManxPower | shiznatix, I doubt you can do that. |
11:21.18 | ManxPower | rkr245, not any different from adding them without an IP address, except you use host=ip.ad.dr.ess instead of host=dynamic |
11:21.24 | ManxPower | ~thebook |
11:21.26 | jbot | methinks thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
11:21.29 | ManxPower | also see the Asterisk book |
11:22.37 | shiznatix | ManxPower, no my dialplan is like this: exten => _95XXXXXXX,1,Dial(Zap/5/{$EXTEN:1}) |
11:22.51 | ManxPower | shiznatix, that is fine. |
11:22.56 | ManxPower | there's NOTHING special about htat |
11:23.00 | shiznatix | ManxPower, but I am using a php script to be able to change the _95XXXXXX and the {$EXTEN:1} |
11:23.02 | rkr245 | o.k |
11:23.07 | rkr245 | thanx |
11:23.19 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
11:23.20 | shiznatix | ManxPower, I want to make sure that the person does not input any characters that asterisk won't use |
11:23.31 | ManxPower | shiznatix, Any time you change extensions.conf you have to issue a reload |
11:23.38 | shiznatix | ManxPower, I know this |
11:24.03 | cj-rm | ManxPower: Being able to simulate certain interactions with asterisk on particular channels and check whether the dialplan/configs behave as expected. If they don't then I'd like to know that the test failed. |
11:24.03 | key2 | !seen kram |
11:24.05 | ManxPower | shiznatix, to be safe only allow a-z, A-Z |
11:24.29 | ManxPower | cj-rm, I know of no such regression testing at this time, but you should ask on #asterisk-dev |
11:24.49 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
11:24.52 | shiznatix | ManxPower, but what about the { and $ and stuff that are used |
11:25.10 | shiznatix | ManxPower, and all the numbers like 9 and 5 |
11:25.10 | ManxPower | so allow those too. |
11:25.39 | shiznatix | ManxPower, but what else is there that can be used is what i am looking to get. what is the full list of stuff that can be used |
11:25.50 | ManxPower | , is a valid character in extensions.conf but if you use it in the wrong place bad things happen |
11:26.27 | brif8 | What causes "ss_thread: CallerID feed failed: Success" and "ss_thread: CallerID returned with error on channel 'Zap/4-1'" ? When I call in on a PSTN line I get this and callerid is blank. using a std phone I see the caller ID information ? |
11:26.33 | ManxPower | So is @ but you don't want to allow people to enter exten => _123@bob.com,1,Dial(Zap/5/whitehouse) |
11:27.10 | ManxPower | brif8, the rxgain is too high or too low |
11:27.29 | brif8 | ManxPower: rxgain = 0.0 txgain = 0.0 |
11:27.42 | brif8 | what would you suggest ? |
11:27.54 | ManxPower | brif8, I suggest you try different values for rxgain |
11:28.08 | ManxPower | increase by 2 at a time or decrease by 2 at a time |
11:28.23 | brif8 | ok I'll first take it up and then try down. thanks |
11:28.32 | cj-rm | Does anyone know how to terminate an internal Local/extension which is running? soft hangup Local doesn't seem to work |
11:29.18 | ManxPower | shiznatix, a better way is to not allow them to enter ${EXTEN:1}, but ask them how many digits to strip and build the ${EXTEN:1} for them |
11:29.52 | ManxPower | cj-rm, it will terminate when it falls off the end of the dialplan or when the physical channel is hungup |
11:30.25 | grem_lin | To be able to use the System() application is there a module that I need to load? |
11:30.31 | ManxPower | greendisease, no |
11:30.42 | ManxPower | grem_lin, not that I know of. Why? |
11:30.43 | shiznatix | ManxPower, I thought about that but what if they want all calls that meet a certain criteria to goto one number |
11:31.05 | grem_lin | I'm getting the message No application 'System' for extension in the console |
11:32.10 | ManxPower | [root@pbx-1 bin]# asterisk -rx "show modules" | grep -a system |
11:32.10 | ManxPower | app_system.so Generic System() application 0 |
11:32.40 | ManxPower | unless you are doing something really stupid with /etc/asterisk/modules.conf app_system.so will autoloafd |
11:33.33 | grem_lin | I'm probably doing something really stupid, I'm specifying the modules I want to load rather than using autoload |
11:33.44 | ManxPower | that's stupid. |
11:33.55 | ManxPower | specify the modules to NOT load. |
11:34.04 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
11:34.25 | puzzled | morning all |
11:34.38 | ManxPower | grem_lin, http://pastebin.ca/index.php |
11:34.41 | ManxPower | hello puzzled |
11:35.08 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
11:35.25 | cj-rm | ManxPower: It's in an infinite loop, and it's a Local extension so there is no physical channel to hangup |
11:35.31 | grem_lin | Thanks for all of your help, ManxPower - I'll do that :) |
11:35.41 | ManxPower | cj-rm, no way other than to stop the PBX that I know of |
11:36.02 | ManxPower | you have to be VERY careful about loops |
11:36.14 | puzzled | hey ManxPower. how's life? |
11:36.14 | ManxPower | that's why I normally put in a loop counter. |
11:36.24 | ManxPower | puzzled, Customers are morons |
11:36.29 | vgster | does i option not work? i am including 2 contexts which have an i option in but when i use the call context i am using it is using gthe i options from the first context it comes across with an i option |
11:36.44 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
11:36.54 | ManxPower | vgster, which of the 5 or 6 i options are you referring to? |
11:37.02 | vgster | invalid |
11:37.13 | shiznatix | Can anyone help me with my zapata.conf configuration. Im having some strange problems with it |
11:37.16 | ManxPower | you mean the "i" EXTENSION |
11:37.20 | vgster | yes |
11:38.09 | ManxPower | vgster, the special extensions like i always have a higher precidence in the local context than in include => contexts. |
11:38.22 | ManxPower | In fact, I didn't even think they worked in an included context |
11:38.25 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
11:38.46 | vgster | hmmm |
11:39.47 | vgster | i was hoping that wasnt the case and I could special a invalid option handler per context |
11:39.54 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
11:40.23 | vgster | any alternatives? |
11:40.59 | *** join/#asterisk cced3 (n=dev2003@222.33.36.205) |
11:41.38 | backblue | damm x100p does not sends any callerid. |
11:42.25 | ManxPower | backblue, SEND? |
11:42.41 | ManxPower | No analog FXO card will send Caller*ID |
11:42.53 | backblue | receive |
11:42.54 | backblue | sorry |
11:43.00 | backblue | :x |
11:43.04 | ManxPower | What country? |
11:43.11 | backblue | portugal |
11:43.31 | ManxPower | The X100P only supports USA FSK Caller*ID |
11:43.54 | ManxPower | one of the many reasons Digium has not sold that card in several years |
11:44.34 | backblue | but there are patchs for germany and uk |
11:44.44 | backblue | or options for zapata.conf |
11:45.31 | ManxPower | backblue, Are you sure those are for the X100P? |
11:45.49 | backblue | ManxPower: yes. |
11:45.53 | ManxPower | and does your country use the same Caller*ID method at DE and UK? |
11:46.37 | backblue | i dont know, but that i think depends on the providers equipment, and not the country! or i'm wrong? |
11:49.19 | *** join/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
11:49.43 | ManxPower | backblue, most providers in a country use the same protocols |
11:50.07 | Seyr | When your on a call and press "#1" to do a transfer (from features.conf), where would the digit timeout be set? for the time between # and 1? |
11:51.02 | backblue | ManxPower: i hope so! :D |
11:51.25 | ManxPower | Seyr, in features.conf |
11:51.52 | backblue | maybe this clone card, does not suport callerid. |
11:51.55 | Seyr | doh! |
11:51.58 | Seyr | thanks ManxPower |
11:53.23 | sternn | exit |
11:53.32 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
11:54.41 | cced3 | who is in CHINA? |
11:58.15 | Hmmhesays | trick question? |
11:58.19 | Hmmhesays | chinese people, duh |
11:58.49 | kaldemar | who _are_ would be better. |
11:58.54 | Zhadnost | cced3> Did you look on #asterisk-dev ? |
11:59.47 | *** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-94.dsl.pltn13.pacbell.net) |
12:01.26 | Hmmhesays | today is going to suck |
12:02.36 | Zhadnost | it already does |
12:03.02 | Hmmhesays | i have a chick upstairs I have to take through 50F weather on a motorcycle |
12:03.06 | Hmmhesays | she will not be impressed by that |
12:03.38 | X-Rob | Depends what sort of bike it is |
12:03.49 | Hmmhesays | a bicycle |
12:03.50 | X-Rob | if it's a harley, she's not gunna be impressed anyway. |
12:03.58 | X-Rob | pushbikes are cool |
12:04.03 | Hmmhesays | i joke |
12:04.12 | Hmmhesays | it's an m50, we went out riding last night and ended back here |
12:04.29 | Hmmhesays | she might be under the impression that I actually have a car right now |
12:05.32 | Ahrimanes | how did she get that impression? |
12:05.39 | Hmmhesays | logical conclusion |
12:05.43 | Ahrimanes | ah |
12:05.48 | Hmmhesays | nice bike, decent home |
12:05.55 | Ahrimanes | she's a girl tho.. so.. logical?! |
12:05.59 | Hmmhesays | lol true |
12:06.19 | X-Rob | Hmmhesays, m50? A scooter? |
12:06.26 | Hmmhesays | suzuki |
12:06.34 | X-Rob | Heh |
12:07.05 | Hmmhesays | http://motorcyclecruiser.com/newsandupdates/blvd-m50-oar-xl.jpg |
12:07.38 | X-Rob | http://www.montesa.hpg.ig.com.br/ima/t315R2001.jpg |
12:07.47 | X-Rob | Oooh |
12:07.50 | X-Rob | that's so not a scooter. |
12:07.59 | X-Rob | it's a harley wannabe |
12:08.10 | Hmmhesays | that bike don't wanna be a harley |
12:08.20 | Hmmhesays | fuel injection, computer controlled timing, shaft drive |
12:08.41 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
12:08.52 | X-Rob | http://www.xaraxone.com/FeaturedArt/ps/assets/images/Honda_Firestorm.jpg |
12:08.53 | Hmmhesays | that bike surpasses harley in every way except namesake |
12:09.01 | X-Rob | 200kph in 3rd gear. |
12:09.03 | X-Rob | I win |
12:09.03 | X-Rob | 8) |
12:09.16 | Hmmhesays | a 1 cylinder bike ? |
12:09.18 | Hmmhesays | yeahright |
12:09.18 | [TK]D-Fender | Nice donor-cycle..... |
12:09.25 | X-Rob | V-Twin, 1000cc |
12:09.46 | X-Rob | [TK]D-Fender, oddly enough, it's the cruisers that have FAR more accidents than the sportsbikes |
12:09.58 | Hmmhesays | not around here |
12:09.59 | X-Rob | possibly because the oldies tend to own the cruisers in .au |
12:11.00 | X-Rob | ours is slightly more custom than that |
12:11.14 | X-Rob | http://www.gladstonewireless.net/users/taz/mybike.html |
12:17.00 | shiznatix | hey can anyone help me with my zapata.conf file? I am having some troubles getting Zap to work properly |
12:17.16 | Zhadnost | it's vaguely possible, go ahead |
12:17.27 | Zhadnost | have you read the voip-info zapata.conf page? |
12:17.47 | shiznatix | Zhadnost, yes I read it several times |
12:18.15 | Zhadnost | did you configure zaptel.conf first and run ztcfg ? |
12:18.59 | shiznatix | Zhadnost, here is my problem. When I try to dial out from my SIP phone through Zap it does not ring the destination phone but instead just says that the call was answered and it gives me this weird high pitched beeping |
12:19.07 | *** join/#asterisk NirS (n=NirS@62.90.49.98) |
12:19.11 | NirS | hello all |
12:19.25 | Zhadnost | weird, anything thin the log files? |
12:20.55 | shiznatix | Zhadnost, not really. here is my stuff: http://pastebin.com/653321 |
12:22.11 | NirS | anyone has an idea why L flag and S flag using dial application won't work ? |
12:22.54 | NirS | anybody home ? |
12:24.09 | ManxPower | 1200 ft of 24 strand multimode fiber. must. resist. bidding. |
12:25.05 | Zhadnost | sounds like mega-money |
12:26.01 | *** join/#asterisk fuzzbawl (i=fuzzbawl@69.44.167.80) |
12:26.03 | shiznatix | Zhadnost, any ideas on my Zap situation? |
12:26.36 | Zhadnost | but then I'm not that experienced with Zap |
12:27.32 | iCEBrkr | *** glibc detected *** double free or corruption (fasttop): 0xf6411f58 *** |
12:27.37 | iCEBrkr | Grrrr |
12:28.08 | ManxPower | shiznatix, analog ports are considered answered as soon as dialing is finished. |
12:28.45 | shiznatix | ManxPower, how do I stop that? how do I make it wait so it will actually ring my the outside line and wait for a answer? |
12:29.10 | ManxPower | shiznatix, you cannot have it wait for an answer on analog ports. I don't know why your calls are failing. |
12:29.22 | ManxPower | you should hear ringing from the telco |
12:30.25 | shiznatix | ManxPower, there is no ringing anywhere |
12:30.49 | *** join/#asterisk skyhawker (n=skyhawke@a62-216-22-13.adsl.cistron.nl) |
12:30.49 | ManxPower | shiznatix, THAT is the problem. |
12:31.00 | ManxPower | You did not find anything helpful when you searched the mailinglist archives? |
12:31.22 | skyhawker | i have a mitel telelphone and was wondering if i could see which external number somebody calls on the phone .. we have two companies here and need to answer accordingly |
12:31.24 | shiznatix | ManxPower, I don't even know where to begin searching. I don't know what I am looking for |
12:31.52 | Zhadnost | silly question, but is the correct zone loaded in zaptel.conf |
12:31.53 | skyhawker | shiznatix: i have that problem toio |
12:32.11 | ManxPower | try X100P and your country name with site:lists.digium.com at google |
12:32.40 | Ahrimanes | skyhawker: many possible ways.. one would be to set callerid to the external number that was called.. |
12:32.59 | ManxPower | skyhawker, callerid=Company A or callerid=Company B BEFORE each channel=> line in zapata,conf |
12:33.30 | skyhawker | ManxPower : thanks .. i am using a external SIP gateway though |
12:33.35 | Zhadnost | Manx> Is setting callerid mandatory in zapata.conf ? |
12:33.56 | ManxPower | Zhadnost, no |
12:34.00 | NirS | anyone has an idea why the L parameter on Dial and S parameters will not work ? |
12:34.27 | ManxPower | skyhawker, then the calls will come into Asterisk and match and exten => line for that dialed number and you can SetCIDName there. |
12:34.40 | NirS | I'm using 1.2.6 |
12:34.45 | ManxPower | NirS, paste a non-working Dial line that uses those options. |
12:34.50 | NirS | hold on |
12:35.07 | ManxPower | Zhadnost, "show application dial" |
12:35.13 | skyhawker | ManxPower : thanks |
12:35.16 | key2 | how can I call a number and dial DTMF keys for calling a number and dialing an extention ? |
12:35.37 | ManxPower | key2, see the D option to Dial |
12:35.44 | Zhadnost | D(<digits). |
12:35.47 | Zhadnost | in the dialstring |
12:35.54 | ManxPower | Zhadnost, make them look it up |
12:36.00 | Zhadnost | s/<digits/<digits>/; |
12:36.05 | key2 | ManxPower: thx |
12:36.13 | ManxPower | and it's in the OPTOINS string, not the dial string |
12:36.26 | NirS | SIP/972544482826@62.90.49.50|120|S(60)M(ngx1_originator_connect^227^d41d8cd98f00b204e9800998ecf8427e443bb091511f9^972544482826^1^moh_promo1^60)L(60000:30000) |
12:36.48 | Zhadnost | wow, that hurts to read. |
12:36.57 | NirS | sorry |
12:36.59 | NirS | that is the dial string |
12:37.02 | NirS | any idea Manx ? |
12:37.04 | Ahrimanes | almost like reading regex's |
12:37.34 | *** join/#asterisk io_error (n=error@87.236.196.130) |
12:38.09 | ManxPower | NirS, are you setting the variables to specify the file to be played. |
12:38.24 | NirS | no, I'm using the defaults |
12:38.31 | ManxPower | try setting them. |
12:38.35 | NirS | ok, will try that |
12:38.42 | Zhadnost | as a curiousity, can you get spandsp to turn a fax into an email? |
12:39.05 | Zhadnost | or even into a tif file. (after that it'd be simple). |
12:39.08 | tzanger | Zhadnost: yes you can |
12:39.44 | tzanger | tiff2ps and ps2pdf |
12:39.44 | tzanger | then email the pdf |
12:39.44 | tzanger | this is well documented on the wiki and google |
12:39.46 | Zhadnost | no doubt,, I was just curious. |
12:40.09 | tzanger | but good lord is spandsp and libtiff picky... it's not exactly spandsp's fault... libtiff has some *fucked up* releases for faxing |
12:40.19 | shiznatix | ManxPower, There was nothing helpful there. I found one guy who has the same problem but it was not answered :( |
12:40.36 | znoG | Zhadnost: you should look into iaxmodem. Great solution and you can use HylaFAX which does just that (converts to PDF and emails) |
12:40.40 | io_error | blah...I can't call in or out on FWD... but I seem to be registered OK. Where do I start on this? |
12:40.45 | tzanger | yes I have to play with iaxmodem again |
12:40.54 | tzanger | I hear redder86 and coppice got 14k4 working |
12:41.22 | Zhadnost | znoG> I already use them together, seems pretty good, never configured asterisk to send emails instead of faxes before. |
12:41.35 | Zhadnost | znoG> been using hylafax for years. |
12:41.48 | tzanger | I want to get my old Ascend Max plugged into hylafax |
12:41.57 | tzanger | telnet to port 5000 or 9000 and you hit a modem |
12:42.05 | tzanger | I just need to write a simple telnet client for hylafax |
12:42.14 | tzanger | there are dozens of telnet SERVERS to serial, but nothing reverse |
12:42.15 | austinnichols101 | ~SER |
12:42.25 | jbot | i guess ser is Sip Express Router - see http://www.iptel.org/ser/ |
12:42.25 | znoG | Zhadnost: oh, you wanted to know if spandsp can do it.. well spandsp receives into tif, so then you run a script on the "h" extension to turn it into pdf and email. It's pretty simple. |
12:42.29 | ManxPower | Yay! I finished adapting my voicemail outcall stuff to be 1.2 specific (more or less) |
12:42.31 | NirS | Manx, that didn't help |
12:42.39 | tzanger | ManxPower: what you say? |
12:43.02 | tzanger | SELECT a.red_mbox as "Property", to_char(sum(b.red_ctime - c.red_ctime),'MI:SS') as "Time Spent", cdr.clid as "Caller*ID" |
12:43.05 | tzanger | FROM realestate_detail a, (select red_uid, red_mbox, red_ctime from realestate_detail WHERE |
12:43.08 | tzanger | <PROTECTED> |
12:43.11 | tzanger | <PROTECTED> |
12:43.14 | tzanger | <PROTECTED> |
12:43.17 | tzanger | WHERE a.red_uid = b.red_uid |
12:43.19 | ManxPower | tzanger, my users are too stupid to have the system send notification of new voicemails via SMS to their cell phone, so the system has to call them to tell them they have new voicemail |
12:43.20 | tzanger | AND a.red_uid = c.red_uid |
12:43.22 | tzanger | AND b.red_uid = c.red_uid |
12:43.24 | tzanger | AND b.red_mbox = c.red_mbox |
12:43.27 | tzanger | AND a.red_mbox <> '?' |
12:43.29 | tzanger | AND cdr.uniqueid = a.red_uid |
12:43.32 | tzanger | GROUP BY a.red_uid, a.red_mbox, cdr.clid |
12:43.35 | tzanger | oh shit |
12:43.37 | tzanger | wrong window guys, sorry |
12:43.40 | tzanger | ManxPower: :-) |
12:43.42 | [TK]D-Fender | tzanger : Sure you're going slow, but its obnoxious! Pastebin! |
12:43.48 | iCEBrkr | tzanger: Hey! thanks man! |
12:43.52 | tzanger | [TK]D-Fender: that was a mistake, sorry :-) |
12:43.57 | tzanger | doing some real estate crpa |
12:43.59 | tzanger | er crap |
12:44.15 | *** join/#asterisk brockj49464 (n=brockj49@41.105.dhcp.hope.edu) |
12:44.17 | austinnichols101 | ~siproxd |
12:44.30 | tzanger | that was supposed to be basted into my pgsql window, not this one |
12:45.29 | key2 | Zhadnost: once the answer machine I call has answered, it does,'t Dial the extension, do you have more info on what I should look for ? |
12:45.50 | NirS | Manx, setting the variable didn't do any help |
12:46.19 | NirS | is it possible that if the dialing activated a Macro to pass the called user into a meetme room, then the L and S parameters can't be used ? |
12:46.45 | shiznatix | Is there anyone that can help me with my Zap problem? |
12:47.05 | NirS | shiz, maybe I can |
12:47.06 | NirS | shot |
12:47.57 | shiznatix | NirS, alright well I am trying to use Zap to call outside lines. I have everything setup but when I call from the SIP phone it just automatically answers and gives me these high pitched beeps |
12:48.06 | shiznatix | NirS, this is the output and my setup: http://pastebin.com/653321 |
12:48.47 | ManxPower | shiznatix, ignore the answer |
12:49.31 | Zhadnost | key2> you may need to write a macro to do a pause, then dial the extra extension. |
12:49.48 | Zhadnost | key2> I've seen stuff liekt his on the mailing list but I'm damned if I can remember how it was done. |
12:49.55 | key2 | Zhadnost: I do a wait |
12:50.07 | shiznatix | ManxPower, ok ill ignore the answer but whats up with this horrible beeping and the failure of the dialed phone to ring |
12:50.10 | *** join/#asterisk Hali_303 (n=surfk@dsl51B6E6EB.pool.t-online.hu) |
12:50.14 | Hali_303 | hi |
12:51.11 | ManxPower | shiznatix, I CANNOT help you with BRI ISDN issues |
12:51.15 | *** join/#asterisk Cheetah (n=Snak@62.217.48.111) |
12:51.21 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
12:51.22 | ManxPower | I thought you were running an X100P |
12:51.29 | NirS | do you also have an E1/T1 card in your box ? or just a TDM 400 ? |
12:51.48 | Cheetah | heya |
12:51.59 | key2 | Zhadnost: that's the D<> that * doesn't like, what docs should I look for getting the syntax on Dialing DTMF during the call? |
12:52.10 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
12:52.10 | shiznatix | ManxPower, ok no problem because this card is not a BRI ISDN card. its a zapata card |
12:52.19 | brif8 | I'm getting can't locate Asterisk/AGI.pm how do I install this using gentoo's emerge ? |
12:52.44 | ManxPower | shiznatix, then why do you have signalling=euroisdn? |
12:52.44 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
12:52.50 | ManxPower | brif8, I don't know if you can. |
12:52.56 | shiznatix | ManxPower, the BRI ISDN card is another card I have installed on the asterisk box |
12:52.57 | Cheetah | What is the better decision for a 30-user phone upgrade? Linksys SPA942 or SNOM 360? |
12:52.57 | ManxPower | just download it and install it |
12:53.04 | Cheetah | anyone has some experiences with those brands? |
12:53.08 | *** part/#asterisk io_error (n=error@87.236.196.130) |
12:53.16 | brif8 | ManxPower: from where ? |
12:53.34 | ManxPower | brif8, search for "asterisk-perl" in Google |
12:53.56 | ManxPower | key2, it's D() not D<> |
12:55.02 | key2 | ManxPower: like Dial(blah,D<2>) ? |
12:55.07 | ManxPower | NirS, It would not suprize me because then you do't have a DIAL |
12:55.15 | key2 | i mean |
12:55.15 | ManxPower | key2, read the damm docs for dial |
12:55.21 | key2 | ManxPower: like Dial(blah,D(2)) ? |
12:55.29 | NirS | Manx, I pasted the options |
12:55.34 | NirS | not the actual dial command |
12:55.36 | *** join/#asterisk fuzzbawl (i=fuzzbawl@69.44.167.80) |
12:55.37 | Cheetah | :'( |
12:56.16 | ManxPower | NirS, is your Dial command secret or something? |
12:56.19 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
12:56.25 | NirS | no |
12:56.27 | NirS | hold on |
12:56.38 | NirS | Dial(SIP/${originator}@62.90.49.50,120,S(${call_limit})M(ngx1_originator_connect^${meetmeroom}^${ivr_session_id}^${originator}^${recording}^${moh_class}^${call_limit})L(${call_limit}000:30000)) |
12:57.13 | Hali_303 | is there a hacker's guide for asterisk? what I mean is something like an introduction of the source code, internal mechanisms, etc. |
12:57.26 | ManxPower | NirS, When you look at the console you see the VALUES of the variables, not the actual ${meetmeroom} string, right? |
12:57.44 | NirS | I see the actual values |
12:57.50 | ManxPower | NirS, I think macros are run before Dial actually happens so I doubt that the call limits apply to the macro. |
12:58.24 | ManxPower | NirS, does everything work EXCEPT the call time limits? |
12:58.28 | NirS | The macro actually is run after the call is answered, and that really happens |
13:00.08 | *** part/#asterisk sercz (n=serz@i3ED6F067.versanet.de) |
13:00.29 | *** join/#asterisk sercz (n=serz@i3ED6F067.versanet.de) |
13:01.20 | ManxPower | NirS, Perhaps it's a string length limitation. Try putting the limit options before the M() option |
13:02.29 | NirS | Hmmm... interesting |
13:02.30 | NirS | will try |
13:03.14 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
13:03.23 | PakiPenguin | hi there , how are you guys today? |
13:03.26 | Cheetah | folks, where (except google) can I ask/find some users who have experience with snom 360 and/or linksys products? |
13:03.44 | Cheetah | we are switching from our old ISDN phone box to VOIP |
13:03.44 | ManxPower | NirS, I encountered a string length problem when doing group voicemail |
13:03.59 | Katty | morning. |
13:04.31 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
13:04.48 | Ariel_ | Morning everyone |
13:07.47 | mitcheloc | good morning |
13:09.49 | Ariel_ | morning hope your day is start great |
13:09.54 | Grizzy | Grizmorning. |
13:10.17 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
13:10.28 | ManxPower | Where are dialplan labels documented? |
13:11.12 | Katty | Ariel_: i'm caffeinating....that's about as far as i've gotten thus far ;) |
13:12.20 | Katty | Grizzy: my legs are killing me :< |
13:12.28 | Hmmhesays | well that went suprisingly well |
13:12.37 | Katty | Hmmhesays: make my legs stop hurting. |
13:12.59 | brif8 | I'm using teliax as my outgoing LD provider (with SIP connection). (1) if I SetCallerID() then I get a busy signal. (2) if I don't then it reports as Denver convention center, which I'm not. Any ideas why ? |
13:13.12 | Katty | Hmmhesays: thanks. |
13:13.31 | Katty | gotta stop doing so many lunges. |
13:13.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:14.01 | Ariel_ | brif8, sounds like they don't allow you to change the callerID |
13:14.05 | *** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net) |
13:14.50 | PakiPenguin | hi , i have a t1 ( sangoma ) and an astribank connected to a server , when i try ztcfg -vvvvv , it gives me this error , can someone tell me what can be wrong please ? http://pastebin.com/653419 |
13:15.20 | x86 | hi PakiPenguin :) |
13:15.34 | brif8 | Ariel_: why would it show incorrectly have they missed something or have I |
13:15.48 | grem_lin | Hey, could anyone give me any clues as to why this dial plan doesn't work as intended http://pastebin.ca/49019 , when I dial a number after authenticating I'm simply getting a fast-busy tone. Any help would be greatly appreciated. |
13:16.17 | Ariel_ | brif8, post how you set your callerID on pastebin so we can see it. |
13:16.44 | *** join/#asterisk DiggerDan (n=chatzill@adsl-69-107-142-7.dsl.pltn13.pacbell.net) |
13:17.43 | brif8 | http://pastebin.com/653426 |
13:17.57 | key2 | how do I set the time lenght for the DTMF i dial ? |
13:18.19 | key2 | if i want each dtmf I dial to last 1s for example |
13:18.22 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
13:18.39 | op3r | anybody knows how to improve the quality of the calls on asterisk? |
13:18.51 | ManxPower | op3r, that would depend on what is causing the problem |
13:18.57 | taec | Have you set Quality(high) ? |
13:19.05 | op3r | taec: nope |
13:19.07 | *** join/#asterisk Router19 (i=SMOKEY@modemcable075.195-131-66.mc.videotron.ca) |
13:19.17 | taec | op3r, oh, well it defaults to Quality(very-very-very-poor) |
13:19.20 | ManxPower | key2, I don't think you can for SIP |
13:19.30 | op3r | because we had a problem dialling new south wales yesterday |
13:19.36 | op3r | and the line quality sucks bad |
13:19.37 | op3r | :( |
13:19.45 | ManxPower | the DTMF length is determind by the device that interfaces with the telco |
13:20.01 | Ariel_ | PakiPenguin, those settings don't look right. since it's telling you that your using the wrong signal |
13:20.15 | key2 | ManxPower: when I do a Dial(tech/chan,,D(123456789)) I only hear one beep, is it normal ? |
13:20.19 | ManxPower | op3r, My car is having problems. How do I fix it?" |
13:20.27 | Katty | with duct tape! |
13:20.40 | ManxPower | key2, no you should hear DTMF for 12345678 and 9 |
13:20.58 | brif8 | Ariel_: do you see any problem with it ? |
13:21.00 | noky | i'm trying to install zaptel in a fc3 follow the wiki...(i have a 2.6 kernel)... but it create a /lib/modules/2.6.9-1.667smp/misc/ztdummy.o |
13:21.13 | syle | lets troubleshoot ManxPower, did you remember to put gas in your car? :) |
13:21.19 | noky | should is not ztdummy.ko ? |
13:21.19 | Ariel_ | just started to read it give me a minute |
13:21.51 | ManxPower | key2, who is your provider? |
13:22.06 | key2 | ManxPower: I do it in internal, just for test purpose |
13:22.28 | key2 | ManxPower: all I want is when I call an extension, it answers and play the dtmf 123456789 |
13:22.33 | ManxPower | key2, Just paste the damn Dial line so I don't have to waste my time asking you for more information |
13:23.27 | key2 | exten => 5665,4,Dial(iax2/key2,,D(123456789)) |
13:23.47 | ManxPower | now what device is "key2"? |
13:23.59 | key2 | my idefisk phone |
13:24.16 | ManxPower | key2, then you need to talk to the idefsk people. |
13:24.38 | key2 | it's not asterisk that generate the dtmf sound ? |
13:24.45 | ManxPower | asterisk sends the messages DTMF 1 DTMF 2 DTMF 3, etc. The idefsk then actually generates the tones and length of tones. |
13:24.59 | ManxPower | noky, asterisk does NOT do inband DTMF with IAX2 EVER |
13:25.11 | key2 | ohh ok |
13:25.37 | Hmmhesays | i see there is some skype to sip software now |
13:25.53 | key2 | but basically, if I want to call an exten, and it answers and play dtmf ? like i did ? |
13:25.54 | noky | ?¿ |
13:25.57 | noky | what? |
13:26.03 | ManxPower | key2, correct. |
13:26.05 | key2 | Hmmhesays ?? where ? |
13:26.23 | Hmmhesays | key2 are you noky |
13:26.24 | ManxPower | noky, you got caught in my "no, asterisk does NOT do inband DTMF with IAX2 EVER" auto complete |
13:26.41 | noky | ok |
13:26.52 | noky | ztdummy.ko or ztdummy.o ? |
13:27.00 | noky | something is wrong |
13:27.13 | key2 | ManxPower: but with this solution, it calls back the phone, it doesnt just answer it and play the dtmf |
13:27.22 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
13:27.28 | op3r | exten => i,1,PlayBack(bad) <---------------- is this the reason why chanspy suck bad? |
13:28.30 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:28.30 | *** mode/#asterisk [+o anthm] by ChanServ |
13:28.46 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
13:29.32 | ManxPower | I guess I should head out to the other office. |
13:30.08 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
13:30.17 | brif8 | Areil_: sorry short power failure |
13:30.18 | NirS | Manx, I changed the order, it didn't do much help |
13:30.22 | NirS | still S and L don't work |
13:31.06 | *** join/#asterisk sorryIdontknow (n=gabriela@200.122.94.137) |
13:31.16 | iCEBrkr | Grrr |
13:31.17 | Hmmhesays | tobad this bullshit runs on windows |
13:31.25 | iCEBrkr | Damnit.. Asterisk keeps dying |
13:31.52 | sorryIdontknow | Hello, I have a question. Can asterisk record/monitor calls by itself, or does in need another program to do that (monitor/mixmonitor)? |
13:32.18 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
13:32.40 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
13:32.56 | NirS | here's something weird |
13:33.06 | NirS | I've brought up the debug level to show debug messages |
13:33.23 | NirS | asterisk doesn't even show anything happen where the time of alert is supposed to happen |
13:33.26 | NirS | which is really weird |
13:33.35 | pdunkel | sorryidontlnow: see res_features it can do it on its own. You usually activate ist with *1 (see features.conf) |
13:33.39 | Ariel_ | brif8, try this: http://pastebin.com/653451 |
13:33.40 | docelm0 | MEW! |
13:33.47 | docelm0 | iCEBrkr, NEWB! |
13:33.50 | sorryIdontknow | Ah, thank you. |
13:34.23 | iCEBrkr | docelm0: U MAK BABBEE JEBUS KRY |
13:34.31 | docelm0 | DAMN RIGHT! |
13:34.33 | iCEBrkr | haha |
13:34.44 | Katty | hey docelm0 |
13:34.54 | docelm0 | whadup? I see you sitting on myspace.. |
13:35.00 | iCEBrkr | I think I may have to upgrade my Asterisk install :( |
13:35.00 | pdunkel | sorryidontknow: Also look at http://www.voip-info.org/wiki-Asterisk+cmd+monitor |
13:35.08 | Katty | i'm trying to wake up |
13:35.09 | *** join/#asterisk Vagabond (n=Vagabond@pdpc/supporter/active/Vagabond) |
13:35.14 | sorryIdontknow | Thanks, pdunkel. :) |
13:36.08 | Vagabond | hey, I'm playing with realtime support, I've got it working for sip peers, but now I'd like to store agents in the database too. I can't find any documentation or discussion about this, is it possible? |
13:36.35 | Katty | Vagabond: hi. |
13:36.40 | *** part/#asterisk sorryIdontknow (n=gabriela@200.122.94.137) |
13:36.54 | Katty | Vagabond: so nice of you to be /nice/ instead of just waltzing in and asking |
13:38.01 | iCEBrkr | Damnit, I need another 727 DID |
13:38.02 | iCEBrkr | c/lear |
13:38.15 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
13:38.22 | [TK]D-Fender | Katty : Potentially preferable to "Why doesn't my A@H work? I put the CD in and EVERYTHING!" ;) |
13:38.28 | Samoied | Hello all! |
13:38.30 | tzanger | [TK]D-Fender: haha |
13:38.51 | Vagabond | so I'm expected to socialize before I get to ask a question? ;) |
13:38.58 | tzanger | [TK]D-Fender: I managed to get my ip501 rollout (12 phones) fully ftp-provisioned this weekend |
13:38.59 | Samoied | Anyone knoe the possible values to _ALERT_INFO for grandstream Handytone? |
13:39.04 | znoG | well, considering the people in here are humans, it's not a bad idea Vagabond. |
13:39.08 | noky | how can i know if ztdummy is OK? i'm trying to configurate the conference (meetme) and i can't... i'm following the wiki for the installation of zaptel... |
13:39.15 | Samoied | I have tried Bellcore-r1,r2, without success |
13:39.22 | [TK]D-Fender | tzanger : Whats scary is just how many of those we get here... Should ask Russell to "enhance" the topic list of "outlawed" junk in here ) |
13:39.25 | noky | what could be wrong????? |
13:39.29 | Vagabond | yeah, I'm just trying to make some progress with this today, not really idle on company time ;) |
13:39.35 | noky | [TK]D-Fender: help please |
13:39.35 | Ariel_ | Vagabond, it would be nice. But no you don't have to. It's just that people tend to ignor |
13:39.40 | tzanger | not a single thing is configured on the phone. hell even the dhcp server gives a separate ip range and tftp option to the polycom MACs :-) |
13:40.09 | [TK]D-Fender | tzanger : Yeah, they go so fast when you have limitied profile types.... I did my 301 from complete scratch within 15 minutes including SIP upgrades. |
13:40.21 | Vagabond | Ariel_: yeah, I guess I'll hang around in here then, it looks like I'll be doing asterisk work for a while |
13:40.42 | noky | :D |
13:40.43 | [TK]D-Fender | noky : Do you see ztdummy loaded? |
13:40.45 | Ariel_ | Vagabond, ok. I wish I could help you. But I do not use realtime and don't belive in it either. |
13:40.49 | noky | yes.. is it... |
13:40.52 | Katty | Vagabond: well i'm social :P |
13:40.54 | noky | appears in 'lsmod' |
13:41.09 | tzanger | [TK]D-Fender: yes NOW that I know what I'm doing it's easy |
13:41.09 | [TK]D-Fender | noky : and cat /proc/interrupts? |
13:41.20 | tzanger | it took me 2.5 hours to get the fucking DHCP server to match a partial MAC though |
13:41.27 | RoyK | [TK]D-Fender: I heard a good that goes back few years. There was a Word Perfect course, starting with installing WP5.1 from 5 1/4" floppies, so teacher says "Insert disk one and press enter"... done with that he goes "Now insert disk two..", and "Now insert the final floppy.." and a lady in the room raised her hand and commented that there weren't room for more... |
13:41.40 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:41.40 | *** mode/#asterisk [+o anthm] by ChanServ |
13:41.41 | [TK]D-Fender | tzanger : Why bother matching a partial MAC? |
13:41.42 | Ariel_ | tzanger, well at least you got it working. |
13:42.00 | tzanger | [TK]D-Fender: because I wanted a separate pool/dhcp options for them and I did NOT want to match individual MACs |
13:42.01 | [TK]D-Fender | RoyK : Golden Oldie.... |
13:42.03 | tzanger | offhand, you do this |
13:42.10 | tzanger | # matches any polycom hardware (vendor MAC 00:04:f2): |
13:42.10 | tzanger | <PROTECTED> |
13:42.10 | tzanger | <PROTECTED> |
13:42.13 | tzanger | <PROTECTED> |
13:42.14 | tzanger | now how fucking ugly is that |
13:42.15 | russellb | [TK]D-Fender: I heard my name!!! |
13:42.18 | tzanger | Ariel_: indeed |
13:42.23 | noky | mmm |
13:42.38 | [TK]D-Fender | tzanger : I just give all options to all devices on my lans... basically Windows PC's ignore the rest so what the hell... |
13:42.45 | tzanger | actually Ariel_ and [TK]D-Fender do you know offhand what the DHCP options are to tell the polycom phones what the FTP username/pass is? I cannot find that info and I'd like to change the user/pass to tighten up security a little |
13:42.52 | tzanger | [TK]D-Fender: yeah, I didn't want to do that :-) |
13:42.56 | [TK]D-Fender | russellb : !!! |
13:43.16 | Ariel_ | tzanger, I have not done that yet. |
13:43.31 | Ariel_ | but it's something I will oneday get to as I would like to change that my self. |
13:43.58 | [TK]D-Fender | tzanger : sorry, gotta do it at the phone level.. |
13:43.59 | noky | [TK]D-Fender: http://pastebin.com/653473 look this |
13:44.14 | tzanger | simiarly, I cannot figure out how to specify http:// instead of ftp stuff but it doesn't look possible either |
13:44.31 | tzanger | seems like a rather egregious omission on polycom's part... they made it SO nice to provision but tripped right at the finish line |
13:44.31 | noky | please |
13:45.10 | pdunkel | Hi, does anyone know which Event headers for "sip notify" are supported by the Snon190/ElmegIP290 phones ? |
13:45.25 | [TK]D-Fender | noky : I don't see ZTDUMMY in your interrupts list..... modprobe it... |
13:46.26 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
13:46.48 | noky | modprobe ztdummy... and the interrupts is the same.. |
13:46.50 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
13:46.50 | noky | :( |
13:46.55 | [TK]D-Fender | tzanger : http you need to do in the provisioning files the first time and doesn't get passed except in a retreived file later |
13:47.08 | tzanger | [TK]D-Fender: eh? |
13:47.56 | [TK]D-Fender | basically you need to feed a config to it that contains the HTTP provisioning server info so that the NEXT time it boots it goes there instead I believe |
13:48.14 | [TK]D-Fender | tzanger : either way though... I like FTP just fine, thanks :) |
13:49.14 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
13:49.20 | [TK]D-Fender | russellb : Hey, care to add "elaborate the "unspeakables" list a bit in the topic? "AMP/FreePBX/A@H" and any others I've missed.... |
13:50.09 | backblue | russellb: do you know if x100p (md3200) clones, do suport callerid? |
13:50.09 | tzanger | [TK]D-Fender: true enough |
13:50.58 | RoyK | the x100p is stupid and doesn't support anything, but zaptel/libpri supports callerid |
13:51.07 | [TK]D-Fender | russellb : Yeah, add X100P's in there too! ;) |
13:52.41 | tzanger | x100p can so do callerid |
13:52.50 | tzanger | it's not done on the card, but neither is any other digium card's callerid :-) |
13:53.05 | noky | crws-----T 1 root root 196, 0 Apr 11 10:13 ctl |
13:53.10 | noky | /dev/zap/ctl |
13:53.15 | noky | is it ok ? |
13:53.22 | noky | 0 byte? o_O |
13:53.48 | [TK]D-Fender | noky : no clue... I've never successfully installed ZTDUMMY before and really need to.... |
13:53.50 | russellb | is there a #aah? |
13:54.04 | brif8 | Ariel_: thanks it works |
13:54.30 | Ariel_ | russellb, no it's support at freepbx |
13:54.42 | syle | noky do you have ztdummy? |
13:54.42 | noky | mm? |
13:54.46 | noky | yes |
13:54.48 | russellb | so confusing :) |
13:54.51 | noky | i compile zaptel for this |
13:54.59 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support |
13:55.04 | Ariel_ | russellb, aah is just asterisk plus freepbx as an iso |
13:55.20 | syle | so what happens on modprobe ztdummy? |
13:55.28 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
13:55.32 | russellb | Ariel_: yeah, i know |
13:55.45 | tzanger | ljam: my gf likes to play tennis... we should get together so you can show me up :-) |
13:55.56 | pdunkel | russellb : :) |
13:56.12 | ljam | tzanger: like that Seinfeld episode? :) |
13:56.22 | tzanger | no idea |
13:56.26 | noky | syle: i use /etc/init.d/zaptel start |
13:56.52 | noky | and i don't log an error... |
13:57.04 | syle | yeah but what happens when you run modprobe ztdummy? |
13:57.15 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F3E1C.dip0.t-ipconnect.de) |
13:57.33 | noky | wait... |
13:57.40 | syle | don;t use scripts if you have a problem, use them after everything works ok |
13:57.44 | ljam | tzanger: doh -- it's a funny episode -- but yah -- lets play some tennis! it's fun -- only started last summer, so I'm not very good, but I can rally alright. I've had a bunch of people ask me to play tennis this week already |
13:57.45 | noky | i will try again, because the zaptel load automatic |
13:57.49 | noky | ok |
13:57.58 | tzanger | cool. I suck mightily at it |
13:58.05 | ljam | lol |
13:58.20 | ljam | I suck at overhand serves.... finally getting to the point that I can get some in now |
13:58.34 | pdunkel | let's open a new room: #tennis :) Anyone up for it? |
13:58.50 | russellb | ooh, i'll play someone :) |
13:59.20 | ljam | I prefer real-life tennis, not virtual :) |
13:59.25 | russellb | yes, me too |
13:59.30 | russellb | ljam: next Astricon, it's on |
13:59.39 | syle | no offence but what does tennis have to do with asterisk? |
13:59.41 | ljam | russellb: done and done! |
13:59.49 | ljam | what does asterisk have to do with asterisk? :) |
13:59.58 | pdunkel | ljam: Now that was an ace if I ever saw one! :) |
14:00.05 | russellb | lol |
14:00.05 | ljam | lol |
14:00.10 | russellb | we got burned. :( |
14:00.13 | ljam | lol |
14:00.18 | Vagabond | hmm, I've been browing the source of chan_agent.c and I don't see anything about realtime storage, am I to assume that agents cannot be stored in the realtime DB right now? |
14:00.23 | ljam | we? I think he was talking to you |
14:00.30 | [TK]D-Fender | ljam : Tennis is cool because of the number of free municipal courts. A good change of pace and gets your outside. I've taken up vollyball and Tenshin Shoden Katori Ryu (one of the oldest schools of Japanese swordsmanship) lately... I may add squash to that list shortly... |
14:00.41 | russellb | Vagabond: i don't think they can, no |
14:00.45 | *** join/#asterisk bweschke (n=bweschke@66.152.225.74) |
14:00.50 | [TK]D-Fender | ljam : and in VB I still haven't recovered my overhead serve :) |
14:00.59 | russellb | Vagabond: if you search for the word "realtime" and it's not there, then yeah, it's not supported :) |
14:01.03 | Vagabond | russellb: do you know if that's planned at all? Or will I have to hack it in? |
14:01.21 | backblue | russellb: do you know if x100p (md3200) clones, do suport callerid? |
14:01.22 | [TK]D-Fender | EVERYTHING! |
14:01.23 | russellb | sure, it's probably on the list of the 5 billion other things we want to do |
14:01.30 | russellb | backblue: they should |
14:01.38 | pdunkel | syle: see chan_tennis.c (will be checked into source as soon as I have approval) Concept: Transport of voice data via tennis balls! |
14:01.44 | syle | play a real sport like hockey :) |
14:01.45 | Vagabond | russellb: anywhere I could look to see what is planned? would it be on the bugtracker? |
14:01.50 | syle | girls play tennis |
14:01.54 | backblue | doesn't work with me! :( |
14:01.58 | [TK]D-Fender | pdunkel : Packet loss must get expensive ;) |
14:02.02 | ljam | syle: EXACTLY -- girls play tennis |
14:02.12 | noky | syle: i have an error |
14:02.16 | [TK]D-Fender | syle : Sure, you play hockey, we'll take the girls ;) |
14:02.19 | russellb | I don't think there is an existing patch on there, but you could look around. Threre are about 300 open issues so it's hard to keep upp |
14:02.20 | [TK]D-Fender | :D |
14:02.23 | cj-rm | hmm... My Zap groups don't seem to be working properly for outgoing calls.... Dial() appears to dial twice! Any ideas? |
14:02.24 | pdunkel | [TK]D-Fender : Yes but is sponsored by ATP! |
14:02.30 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.47.Dial1.SanJose1.Level3.net) |
14:02.32 | Vagabond | russellb: thanks |
14:02.34 | ljam | [TK]D-Fender: I play beach volleyball in the summer ---mmmmm... summer girls :) |
14:02.41 | noky | syle: http://pastebin.com/653503 |
14:02.48 | russellb | ljam: on your fake beach! |
14:02.49 | noky | i don't know how fix |
14:02.57 | syle | is asterisk running as root? |
14:02.59 | noky | i follow the wiki... i don't understand |
14:03.00 | ljam | russellb: with the fake... you know :) |
14:03.00 | pdunkel | Any takers on my sip notify with snom question? |
14:03.10 | noky | asterisk is doesn't running at this moment |
14:03.14 | noky | should be run? |
14:03.20 | syle | when you start it, as what user? |
14:03.27 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
14:03.38 | noky | root |
14:03.43 | noky | sorry, is running now... |
14:03.45 | noky | root 2090 0.0 0.7 19220 7348 ? Sl 10:24 0:00 /usr/sbin/asterisk -vvvg -c |
14:03.45 | bweschke | Vagabond: what are you looking to do? |
14:03.47 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.47.Dial1.SanJose1.Level3.net) |
14:03.49 | [TK]D-Fender | noky : If you don't see ZTDUMMY loaded, forget it. You sure you uncommented it in the source file and recompiled and everything? |
14:03.57 | noky | yes.. |
14:03.58 | cj-rm | I'm not sure if it detects that my channel is busy |
14:04.08 | pdunkel | syle: While we are at * and users. What are the pros and cons of running * as root? |
14:04.11 | noky | i'm sure |
14:04.49 | noky | in Makefile i uncomment the ztdummy... make clean && make linux26 && make install ... and then (i have use udev) configurate the udev follow README.udev |
14:04.50 | [TK]D-Fender | pdunkel : If there's something exploitable you could comprimise your entire server..... |
14:05.05 | Vagabond | bweschke: I'd like to be able to have agent numbers/passwords in the realtime db so they're easier to add |
14:05.11 | Vagabond | and generally manage |
14:05.20 | syle | so you have your udev entries in /etc/udev/rules.d/50-udev.rules ? |
14:05.29 | ljam | realtime? ugh -- just use a DB :) |
14:05.35 | noky | i test with 'make' instead of 'make linux26'... |
14:05.40 | noky | yes... |
14:05.48 | syle | did you reboot after adding them? |
14:06.00 | noky | yes.. i reboot |
14:06.08 | noky | but /dev/zap/ctl appears with 0 bytes |
14:06.09 | cj-rm | Why is it when I try and make an outbound dial with Zap groups that my extension runs twice?????? |
14:06.16 | noky | i don't think that is ok.. |
14:06.30 | ljam | ok -- off to get some breakfast, then off to implement slony-I :) |
14:06.45 | cj-rm | Yet when I specify the specific channel to call on e.g. Zap/3 then it doesn't. |
14:06.52 | pdunkel | [TK]D-Fender: Yeah, the question is more like what issues are to watch out for if I decide not to run as root as well as likelyhood of exploits. Of course these can only be experience reports and I do know the general issues well. I was hoping for something more along the line of experience reports. |
14:06.59 | syle | make linux26 was suppose to be depreciated anyways in recent code bases , shouldn;t make a difference |
14:07.14 | [TK]D-Fender | cj-rm : pastebin your entire extensions.conf and zapata.conf. |
14:07.39 | syle | its suppose to be 0 bytes |
14:07.44 | noky | ok.. |
14:07.50 | pdunkel | ljam: just modprobe coffe && modprobe cigarette (That does it for me!) |
14:08.10 | [TK]D-Fender | pdunkel : To tell you the trusth I'm not aware of more that 1 occurance where anything serious could happen, its just a "purist approach" to not run daemons as root. |
14:08.12 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
14:08.38 | syle | pastebin 3 things will you, modprobe ztdummy, ls -al /dev/zap , and cat /etc/udev/rules.d/50-udev.rules ok |
14:08.44 | cj-rm | [TK]D-Fender: Can Dial() on a group match a Local channel? |
14:08.57 | pdunkel | [TK]D-Fender: Well I guess I'm about to find out how much of a "purist" I really am. Thanks |
14:09.18 | syle | lsmod as well |
14:09.18 | noky | oks |
14:09.28 | pdunkel | Anyone here has any SIP knowledge (relating to SIP NOTIFY) ? |
14:09.40 | pdunkel | Or are you all ZAP guys ? |
14:10.13 | syle | lol |
14:10.31 | [TK]D-Fender | pdunkel : I wouldn't and don't personally bother... |
14:10.36 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
14:11.04 | noky | syle: http://pastebin.com/653518 |
14:11.06 | pdunkel | *pdunkel sizzles |
14:11.15 | *** join/#asterisk cj-rm (n=cjrm@81-86-30-78.dsl.pipex.com) |
14:11.15 | iCEBrkr | Grrr |
14:11.23 | iCEBrkr | Everyone elses system runs just fine *Grumps* |
14:11.30 | pdunkel | someone give me a raygun so i can zap some as well! :) |
14:12.40 | [TK]D-Fender | iCEBrkr : * is perfectly stable! When it crashes it doesn't move at all! |
14:12.45 | iCEBrkr | haha |
14:12.55 | iCEBrkr | Well, I *Am* dropping 35 call files at a time. |
14:13.07 | pdunkel | iCEBrkr: No mine crashes about 2 a day, so I have implemented a cron check script until I have found out why the crash happens. *pdunkel sends sympathy |
14:13.15 | iCEBrkr | Odd thing is. I just hit my up-arrow once to relaunch asterisk -cvvvvv and it worked. |
14:13.37 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
14:15.31 | noky | syle: something wrong ? |
14:15.35 | *** join/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com) |
14:15.43 | opc0de | hey anyone here use chan_btp ? |
14:15.46 | *** join/#asterisk pjz (n=pj@66.219.59.183) |
14:15.59 | pjz | how is the soundpoint 501 different from the soundpoint? |
14:16.17 | pjz | my autoboot-and-upgrade-firmware stuff doesn't seem to work on the 501 but did on the 500 |
14:16.38 | pjz | the 501 is saying 'Error updating BootROM' |
14:16.43 | opc0de | I'm trying to figure out why "client => user,00:11:22:33:44:55,Zap/4/1234567891" won't dial 1234567891.. it simply picks up Zap/4 and connects me, without dialing then umber |
14:16.49 | pjz | I've got bootrom v2.6.1 |
14:17.07 | tzanger | pjz: hmm |
14:17.14 | tzanger | I didn't try to upgrade my bootrom, I'm using 2.6.1 |
14:17.52 | opc0de | pjz: |
14:17.56 | [TK]D-Fender | Leave your BR alone if at all possible.... |
14:17.57 | opc0de | I've got bootrom 3.1.3 if you want |
14:17.59 | syle | it looks good noky |
14:18.09 | syle | did you make install latest libpri first |
14:18.11 | pjz | I don't really want to go to 3.1.3 since that's all https and stuf |
14:18.17 | opc0de | pjz: ar eyou using ftp or tfp? |
14:18.22 | cj-rm | [TK]D-Fender: http://pastebin.com/653535 |
14:18.22 | pjz | opc0de: ftp |
14:18.30 | cj-rm | [TK]D-Fender: The relevant bits are there. |
14:18.36 | opc0de | pjz: does your ftp log show the phones downloading the software? |
14:18.47 | tzanger | 2.6.1 and sip version 1.6.5.0043 |
14:18.50 | opc0de | no one here uses bluetooth? |
14:18.52 | OliverX | wich ports must i forward in my nat to register asterisk to the sipgate account? |
14:18.58 | noky | libpri ? |
14:19.35 | stoffell | hm, why does the MACaddr-app.log of polycom phone keeps getting written in ftp homedir? (i specified logs/ dir in xxx.cfg) |
14:19.39 | bkw_ | OliverX, if asterisk had a STUN client it wouldn't matter :P |
14:19.49 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
14:19.58 | syle | try modprobe zaptel first |
14:20.01 | syle | same error? |
14:20.37 | noky | mm wait me please |
14:21.10 | opc0de | is there any way to define a channel which by default dials a certain fphone number? |
14:21.17 | noky | modprobe zaptel is ok |
14:21.26 | noky | i don't see some error. |
14:21.27 | opc0de | by using Zap/4/1234567891, it never dials the phone number, it only picks up the channel |
14:21.29 | [TK]D-Fender | cj-rm : I don't see where those variables you use are being set.. include MORE.... |
14:21.37 | pjz | opc0de: hrm, no they're not logging in |
14:21.43 | [TK]D-Fender | cj-rm : Show me everything that toushes it... |
14:21.51 | pjz | opc0de: do 501s have different usernames? |
14:22.14 | cj-rm | [TK]D-Fender: They're being set in the call file which is copied into /var/spool/asterisk/outgoing |
14:22.15 | opc0de | pjz: you've gotta set the username/password in t he admin config screen on the phone |
14:22.43 | [TK]D-Fender | OliverX : typically 5060, 10000-20000, all UDP |
14:22.54 | [TK]D-Fender | cj-rm : Ok, I don't know call-files really... |
14:22.58 | pjz | opc0de: oh, nm, it's logging in correclty |
14:23.20 | pjz | opc0de: oh wow |
14:23.23 | noky | must i wait seconds to modprobe ztdummy? |
14:23.28 | opc0de | pjz: what uid/password are you using right now? |
14:23.32 | pjz | | ProFTPD terminating (signal 11) |
14:23.34 | opc0de | 456/Polycom ? |
14:23.36 | opc0de | that's not good |
14:23.40 | noky | because if i wait some seconds ... the modprobe ztdummy works ok |
14:23.45 | pjz | yeah, my guess is that that's the problem :) |
14:23.46 | noky | after the modprobe zaptel |
14:24.02 | pjz | Sig 11 is Segfault |
14:24.03 | Samoied | Anyone know the possible values of _ALERT_INFO for Grandstream Handytone? |
14:24.08 | syle | is it in lsmod now? |
14:24.13 | syle | ztdummy |
14:24.55 | noky | when i do modprobe zaptel doesn't appear ztdummy |
14:25.08 | syle | is it in lsmod now? |
14:25.10 | noky | but i do modprobe ztdummy... and i don't see error... and now is ok |
14:25.12 | noky | yes |
14:25.25 | greendisease | can someone lastlog me please? |
14:25.37 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
14:25.41 | syle | great well script it to start it like that then, have a good day |
14:25.59 | noky | haha |
14:26.09 | noky | syle: i will try if works |
14:26.19 | noky | thanks for your time buddy |
14:26.50 | syle | i don;t know why you have to load zaptel first but whatever , just do modprobe zaptel;sleep 5; modprobe ztdummy then in some startup script |
14:26.50 | cj-rm | [TK]D-Fender: The variables are set correctly, it works perfectly when the values for ${OUTBOUNDTRUNK} are set to Zap/3 for the 1st Dial() and Zap/4 (for the second). But not when I use the channel group. |
14:27.25 | cj-rm | [TK]D-Fender: By that I mean when ${OUTBOUNDTRUNK} is replaced by Zap/3 and Zap/4 btw... |
14:27.35 | syle | i don;t know what release etc your running, nor do i care but at least its working, glad to help |
14:28.45 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
14:28.47 | vader-- | hello |
14:29.51 | noky | thanks |
14:33.48 | noky | syle: i follow heard in asterisk the operator saying "..please try again..", my extension look good.. and in /proc/interrupts doesn't appear the ztdummy... like say [TK]D-Fender must to appear, not? .... |
14:34.57 | noky | my extension of Meetme |
14:35.33 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
14:35.40 | syle | it will appear on kernel version 2.4 not in 2.6 |
14:36.26 | syle | your concern is it loads and you can see it in lsmod |
14:36.33 | [TK]D-Fender | op3r : which user/pass are you trying to figure out? there is another mixed case one "plcmspip" for FTP which is "default" |
14:36.47 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:36.52 | noky | ok |
14:36.55 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
14:37.31 | noky | do u use realtime extensions syle ? |
14:37.31 | syle | yes |
14:37.42 | noky | | 15 | default | 999 | 4 | MeetMe | 1234|M | |
14:37.44 | syle | be a pain in the ass to interact with asterisk froma webpage otherwise |
14:37.50 | noky | this extension is ok, not ? |
14:37.58 | noky | hehe |
14:38.34 | noky | in meetme.conf only have |
14:38.35 | noky | [default] |
14:38.37 | noky | conf => 1234 |
14:38.52 | brif8 | anyone using a Cisco 7920 (wireless) I keep getting No AP found ?? |
14:38.54 | noky | when i dial 9999 i listen "please try again.." :( |
14:39.03 | noky | 999* |
14:39.45 | noky | the priority 1 is Answer, the priority 2 is a wait 1 second and the priority 3 is a wait 2 seconds... |
14:40.02 | noky | then the priority 4 is Meetme .. and the priority 5 is Hang up |
14:40.08 | noky | must be ok... |
14:40.55 | syle | looks good to me, haven;t played with meetme , so can;t help you there, realtime looks good, check /var/log/asterisk/debug as well for info |
14:41.02 | noky | Apr 11 11:04:40 DEBUG[3949] app_meetme.c: 1234 isn't a valid conference |
14:41.04 | noky | :o |
14:41.08 | noky | appears in log... |
14:41.16 | noky | oks. |
14:43.44 | *** join/#asterisk gandhijee (n=loser@host-66-202-34-162.spr.choiceone.net) |
14:43.56 | gandhijee | anyone know of a softphone that can do IP dialing? |
14:43.57 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
14:44.37 | gandhijee | ?? |
14:45.06 | *** join/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net) |
14:45.08 | syle | why not just use your cell phone, switch out your sim card, put in a gsm gateway |
14:45.21 | syle | hate a phone that is not wireless hehe |
14:46.16 | gandhijee | i c. |
14:46.33 | gandhijee | i have wifi phones |
14:46.48 | syle | kewl, what kind? |
14:46.48 | Lino` | brif8 |
14:46.54 | gandhijee | i am just lookin for a softphone that can do IP to IP calling |
14:46.55 | Lino` | which ap do you have? |
14:47.06 | gandhijee | ZyXEL |
14:47.10 | gandhijee | and i just got 2 linksys WIP300's |
14:47.21 | gandhijee | haven't got to test those yet, they are in VA |
14:47.36 | brif8 | Lino` D-Link 3200AP |
14:47.38 | *** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
14:47.39 | Lino` | hmmm |
14:47.44 | Lino` | you need to broadcast the SSID |
14:47.49 | brif8 | I got it had to set SSID |
14:47.54 | Lino` | you need to set the encryption parameters for the 7920 |
14:48.00 | *** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu) |
14:48.00 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
14:48.09 | Lino` | and you need to allow the phones mac address |
14:48.10 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
14:48.12 | brif8 | now getting Connecting to CallManager 0 (it cycles 0 to 5 and back again) |
14:48.17 | syle | i find desktop phones are only good for speaker phone personally, polycoms are best for that, otherwise use your cell phone or wifi phones, at least you can still take a piss and talk on the phone |
14:48.17 | Lino` | ok |
14:48.20 | Lino` | thats fine |
14:48.26 | Lino` | you have a call manager or asterisk? |
14:48.30 | brif8 | * |
14:48.35 | Lino` | most probably asterisk in this channel *stupid me* |
14:48.41 | jsharp | I hate people who take a piss while I'm on the phone with them. |
14:48.47 | Lino` | now you need sccp |
14:48.50 | Strom_M | syle, sure, if you don't care about the sound of GSM compression |
14:48.54 | Lino` | either chan_sccp2 or chan_skinny |
14:48.56 | syle | call me then, i need to take one :) |
14:49.04 | Strom_M | or if you never have to worry about battery life |
14:49.22 | Lino` | :D |
14:49.32 | file[laptop] | meep meep |
14:49.34 | brif8 | I followed the HOW to SCCP but so far no joy |
14:49.37 | syle | cell phones are battery life, i think they are most popular :) |
14:49.48 | [TK]D-Fender | or range, or risk of being dropped.... |
14:50.00 | syle | then install access points |
14:50.00 | brif8 | Lino`: do you use one and do you have it working ? |
14:50.06 | syle | solution for everything |
14:50.18 | gandhijee | my wifi phones run PCMU |
14:50.32 | Strom_M | basically, mobile and cordless phones are only good for situations where you need to be in many different locations all the time. Otherwise, corded phones are the way to go |
14:50.49 | gandhijee | syle: installin AP's don't get rid of the handoff problem |
14:50.52 | syle | you have no life if thats what you think , no offence :) |
14:51.12 | gandhijee | any of you guys check out RoamAD's software? |
14:51.25 | gandhijee | its pretty cool stuff |
14:51.25 | file[laptop] | Strom_M: are you involved in the... meeting of doom today? |
14:51.32 | brif8 | do I need to get the sccp2 or is sccp fine , how do I know I have the right one ? |
14:51.33 | Strom_M | meeting of doom? |
14:52.07 | Hmmhesays | thank you federal guberment |
14:52.08 | Katty | hey Hmmhesays, i found a pretty girl for ya! |
14:52.12 | Hmmhesays | oh yeah? |
14:52.36 | iDunno | can you find me one too? |
14:52.40 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
14:52.48 | brif8 | Lino`: I have /etc/asterisk/sccp.conf what else is needed ? |
14:52.57 | FlyboySR22 | Has anyone run into any issues running * on a x64 platform...? We are looking at Suns new Sunfier X2100 servers to deploy * on.... |
14:53.15 | syle | waste of money |
14:53.28 | Katty | iDunno: i don't share my girls with strangers :P |
14:53.28 | tzafrir | FLeiXiuS, I know Debian has occasionally some patches... |
14:53.37 | FlyboySR22 | syle, WHy..they are cheap and fast |
14:53.43 | FlyboySR22 | ?? |
14:53.46 | syle | define cheap |
14:53.53 | FlyboySR22 | sub $1K |
14:54.03 | FlyboySR22 | and they fit in with the rest of my sun gear :-) |
14:54.27 | syle | that might be kewl, i;ve always found i could buy 3 intel servers for the price of 1 sun machine |
14:54.43 | FlyboySR22 | white-box server...? |
14:54.57 | FlyboySR22 | I have yet to find a good $300 server |
14:54.59 | syle | 1k is good though if they are that cheap |
14:55.06 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
14:55.12 | gandhijee | Flyboy: then just get a linux and run them in 32-bit mode.... |
14:55.25 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
14:55.25 | iDunno | Katty: awww :( |
14:55.27 | syle | interested to hearing about asterisk in 64 bit code |
14:55.29 | gandhijee | i got my dual core dell for like 900 w/ 2 gigs for ram |
14:55.42 | FlyboySR22 | gandhijee, kind of defeats the purpose of a 64 bit system dosn't it..? |
14:55.43 | gandhijee | *of |
14:55.45 | syle | you have the 64 bit optimizations ready? |
14:55.46 | FlyboySR22 | gandhijee, New..? |
14:56.12 | gandhijee | Flyboy: no, but IRC the zaptel drivers are not ready for 64 bit yet. |
14:56.25 | FlyboySR22 | ah |
14:56.28 | gandhijee | that is if u plan on running any PSTN lines |
14:56.41 | Seyr | syle: the $900 Sun server is running AMD, thats why its so cheap |
14:56.44 | gandhijee | i just dont come by often |
14:56.46 | FlyboySR22 | how about the libpri stuff..? I am only going to run PRI cards... |
14:56.49 | syle | yeah, he;ll be running in 32 bit mode anyways |
14:56.58 | syle | so won;t matter much hehe |
14:57.06 | FlyboySR22 | Opteron is 64 bit I thought..? |
14:57.08 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
14:57.16 | FlyboySR22 | Sun X2100 = Opteron |
14:57.17 | gandhijee | Flyboy: you still need zaptel for the PRI stuff |
14:57.24 | FlyboySR22 | for timing I assume |
14:57.27 | Vagabond | hmm, when you use AgentCallbackLlogin, and it asks for an extension, how do I define one it won't say is invalid? |
14:57.44 | gandhijee | Flyboy: no to make the zaptel channels, i assume you are getting a T1 or E1? |
14:57.52 | gandhijee | as i dunno where u are |
14:58.04 | FlyboySR22 | Yes - sorry. PRI |
14:58.07 | FlyboySR22 | I am in the US |
14:58.11 | FlyboySR22 | San Diego |
14:58.12 | jsharp | You'll need zaptel and libpri. |
14:58.17 | *** join/#asterisk plasko (n=plasko@triana.kmpanilla.com) |
14:58.30 | FlyboySR22 | OK - so stay away from 64 bit for now... |
14:58.41 | gandhijee | ok |
14:58.42 | gandhijee | ya, you still need the Zaptel |
14:58.42 | gandhijee | thats what makes those nice little ZAP/1 channels for you so you can get an incomming call. |
14:58.54 | syle | hell no |
14:59.00 | gandhijee | unless you want to have fun =) |
14:59.03 | syle | i want to hear results :) |
14:59.03 | jsharp | zaptel drives the card. libpri handles layer 3 ISDN signalling. |
14:59.18 | *** part/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
14:59.27 | FlyboySR22 | but zaptel does not work well on 64 bit..? These would be production boxes |
14:59.33 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
14:59.44 | syle | because a box is 64 bit doesn;t mean it can;t run 32 bit code |
14:59.51 | syle | you just adjust your compiler |
15:00.27 | FlyboySR22 | syle, got it, but again, then why get a 64 bit processor if I have to run in 32 bit mode..? |
15:00.32 | brif8 | Lino`: are you using the Cisco 7920 or anyone else ? |
15:00.38 | FlyboySR22 | are there other benefits ..? |
15:00.41 | jsharp | Your kernel drivers have to be 64-bit clean otherwise bad stuff happens. |
15:00.41 | syle | well that was my point, not really needed |
15:00.54 | syle | i was just interested to hear how it works for you |
15:00.56 | FlyboySR22 | syle, :-) Got it |
15:01.37 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
15:01.40 | syle | if your running production yeah i wouldn;t chance it personally |
15:01.41 | brif8 | I have sccp loaded but it still seems that the 7920 won't connect to * ? |
15:02.13 | *** join/#asterisk gandhijee (n=loser@host-66-202-34-162.spr.choiceone.net) |
15:02.28 | FlyboySR22 | syle, Thanks for the informaiton, I appreciate it. I have not done anything with a 64 bit processor yet so was looking at trying it out |
15:02.36 | Hmmhesays | so who do ya'll register your domain names through |
15:02.53 | syle | i thought you had lots of SUN equipment, they are usually all 64 bit these days hehe |
15:03.04 | Katty | we use register.com |
15:03.06 | Katty | but don't use it |
15:03.06 | FlyboySR22 | yes, but they are running Solaris |
15:03.07 | gandhijee | The Sparc boxes are |
15:03.11 | Katty | they don't support reverse dns lookup |
15:03.11 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F3E1C.dip0.t-ipconnect.de) |
15:03.12 | Hmmhesays | what do you run on them syle, I have a sparc 3000 doing nothing |
15:03.20 | FlyboySR22 | we have the E450s and some of the Netras |
15:03.22 | jsharp | sparc 3000? |
15:03.28 | syle | i;ve only run solaris on sparcs |
15:03.30 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
15:03.33 | FlyboySR22 | we got rid of the E250s |
15:03.33 | gandhijee | x86 netras? |
15:03.46 | syle | your big ass fiber array channels and veritas |
15:03.49 | FlyboySR22 | not sure, I would have to ask one of my Sun guys |
15:05.05 | syle | SUN is stable, never had a problem with it ever |
15:05.21 | syle | i just couldn;t justify the cost is all |
15:05.31 | jsharp | Yah, you pay for the stability. |
15:05.53 | FlyboySR22 | Netra T1 = Sparc 64 bit according to my Sun guy |
15:05.57 | syle | yeah well you can argue you pay your sys admin for stability |
15:06.03 | syle | so whatever works :) |
15:06.16 | FlyboySR22 | :-) |
15:06.20 | FlyboySR22 | how true is thata |
15:07.59 | brif8 | anyone using the 7920 where did I goof. I keep getting (Connecting to CallManager 0) it cycles 0 -> 5 |
15:08.28 | brif8 | I have sccp loaded and sccp show devices and channels are emty version is 20060408 |
15:08.52 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
15:09.10 | jsharp | Sounds like your sccp.conf isn't right. If show devices is empty, there are no devices properly configured. |
15:09.14 | X-Rob | Grrrr. |
15:09.30 | X-Rob | Any way in the dialplan, that anyone knows of, to check if an exten => 1234 exists? |
15:10.17 | tzanger | X-Rob: that's a damn good question. Short of a System(grep...) I odn't think there is a way |
15:10.25 | pdunkel | X-Rob: Do you mean you want to check with a dialplan app whether an ext exists? |
15:10.30 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
15:10.58 | X-Rob | pdunkel, I want to GotoIf (${exten_exists(1234)}?1234:fail) |
15:11.11 | Hmmhesays | damnit |
15:11.20 | pdunkel | If that's what you mean, you coud do a TryExec(Goto(<context>,<exten>,<priority>) as of the current svn trunk |
15:11.44 | X-Rob | pdunkel, bugger. Can't use trunk yet, as I want this to work in 1.2 |
15:11.47 | pdunkel | After that you can check ${TRYSTATUS} for FAILED or SUCCESS |
15:12.23 | tzanger | X-Rob: offhand, why do you need this? |
15:12.28 | pdunkel | (This is something I submitted) I can giv you the patch. I have it working against 1.2.4 and 1.2.6 without a problem! |
15:12.39 | X-Rob | tzanger, routing based on Zap channel names. |
15:12.51 | X-Rob | I've got it _almost_ working. |
15:13.04 | Hmmhesays | how could someone waste a good domain name like lostpacket.org |
15:14.24 | X-Rob | The issue is, I have to do it in a macro.........ooh, you know, I might NOT have to |
15:14.35 | X-Rob | no |
15:14.36 | X-Rob | bugger |
15:14.37 | X-Rob | I do. |
15:14.58 | X-Rob | I don't want to have to set all the zap channels to a different context |
15:15.03 | pdunkel | X-Rob: You want the replacement for app_exec.c? |
15:15.20 | X-Rob | pdunkel, I can't change asterisk - we've got enough dependancies as it is 8) |
15:15.35 | X-Rob | (this is for freePBX 2.1) |
15:15.57 | pdunkel | X-Rob: Ahh. Well suit yourself :) |
15:16.16 | RoyK | ~disclaimer? |
15:16.17 | jbot | I disclaim all of you!, or "fortune -m 'Void where'" |
15:16.47 | RoyK | ~jbot? |
15:16.48 | jbot | somebody said jbot was only marginally useful at best, He got a C- on his Turing Test |
15:17.07 | *** part/#asterisk sercz (n=serz@i3ED6F067.versanet.de) |
15:17.22 | X-Rob | how the smeg am I going to do this |
15:17.54 | noky | syle, [TK]D-Fender: it's work! |
15:18.00 | iDunno | magic. |
15:18.01 | X-Rob | I think I might just go 'if you're routing via channel, then you can't route via Caller ID. Accpet this and move along' |
15:18.28 | Hmmhesays | wow godaddy is cheaper than registerfly |
15:18.38 | [TK]D-Fender | noky : congratulations |
15:19.02 | pdunkel | X-Rob: The problem is that Goto returns -1 if it can't find the exten. This tells * to stop dialplan execution. There really is no way aound this. I was laboring over the same problem. That's why I came up with this new app. |
15:19.11 | syle | goddaddy is run in USA though so they shut down domains as they see fit |
15:19.27 | syle | offshore is abit more secure |
15:19.32 | X-Rob | Hmmhesays, you're not _serously_ considering using godaddy after what they did to me, are you? |
15:20.20 | syle | did they make you their bitch? |
15:20.27 | X-Rob | they fucked my arse well. |
15:20.41 | syle | lubed or 12 inches dry? |
15:20.48 | X-Rob | 'We've got all your data, and all your backups. Fuck off. Any questions? email legal@godaddy.com' |
15:21.15 | syle | oww you hosted with them |
15:21.18 | X-Rob | Yup. |
15:21.33 | syle | i don;t think i;d ever host without colocated box |
15:21.37 | pdunkel | There probably are ladied present. goto #inappropriate or #fun for some #fucking :) |
15:21.49 | noky | thanks [TK]D-Fender & syle !!! |
15:21.52 | noky | :D |
15:22.38 | *** join/#asterisk salviadud (n=ralfalfa@201.135.13.124) |
15:22.51 | pdunkel | *pdunkel just slipped on it :) |
15:23.01 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
15:23.06 | key2 | what is the name of the zaptel monitoring tool ? |
15:23.40 | syle | cd /usr/src/zaptel |
15:23.48 | syle | all your binaries are there |
15:23.56 | a1fa | all your binz belong to us |
15:24.04 | mitcheloc | * are belong to us |
15:24.28 | Katty | what a bunch of oddballs. |
15:24.30 | mitcheloc | a1fa: personally, i think the channel logo topic should be "all your pbx are belong to us" |
15:24.41 | a1fa | Katty, hi sweetie |
15:24.52 | a1fa | hehe |
15:24.54 | a1fa | then change it |
15:25.01 | syle | Kattie likes being center of attention so tell her how nice her tits look today :) |
15:25.04 | a1fa | anybody experiencing odd issues with 1.2.6 and sip? |
15:25.13 | a1fa | my pbx picks up calls on random |
15:25.13 | Katty | wow, syle really doesn't know me |
15:25.20 | Katty | that's kinda sad. |
15:25.29 | salviadud | don't be dissin' on Kat |
15:25.36 | a1fa | salviadud : i love me some katty |
15:25.49 | pdunkel | a1fa: Yes, I keep having open channels that never go away. |
15:25.49 | salviadud | she's funny |
15:26.02 | syle | salviadud how much do you like that cock sucking you do :) |
15:26.03 | Katty | i just randomly drop by and say something and disappear again |
15:26.03 | salviadud | she's got some funny pics in her site |
15:26.04 | a1fa | she is hot |
15:26.09 | a1fa | katty is hot |
15:26.11 | a1fa | woom |
15:26.15 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
15:26.16 | X-Rob | Sigh. |
15:26.17 | Katty | oh for goodness sake |
15:26.19 | Katty | grow up |
15:26.22 | a1fa | haha |
15:26.27 | a1fa | everbody msg katty -> asl |
15:26.30 | Katty | i am not a slab of meat :P |
15:26.38 | salviadud | syle, i'm not gay... |
15:26.38 | syle | of course you are, get use to it |
15:26.41 | a1fa | i believe katty has enum too :) i am sure we can peer to her :P |
15:28.03 | syle | salviadud thats to bad, i know a guy that was asking about you |
15:28.54 | salviadud | riiiiight |
15:29.31 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
15:32.33 | Lino` | oh |
15:32.44 | DoktorGreg | <PROTECTED> |
15:32.52 | DoktorGreg | my pri card gets here tomorrow |
15:33.22 | a1fa | good for you gay |
15:33.25 | a1fa | :) |
15:33.27 | a1fa | ehhe |
15:34.22 | a1fa | anybody using 1.2.6 and broadvoice? |
15:34.47 | [TK]D-Fender | a1fa : A company I consult at. |
15:34.52 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
15:35.03 | syle | i;m surprised their still around |
15:35.10 | a1fa | [TK]D-Fender : hehe.. asterisk wont pickup inbound calls |
15:35.15 | *** join/#asterisk tdonahue-laptop (n=tdonahue@208.51.101.201) |
15:35.16 | a1fa | it worked fine in 1.2.5 |
15:35.25 | a1fa | i see a call come in, it rings, and it rings out |
15:35.29 | a1fa | SIP Debugging Enabled for IP: 147.135.0.128 |
15:35.33 | a1fa | nothing in the log file |
15:35.44 | a1fa | or debug |
15:36.05 | a1fa | Expiry for sip.broadvoice.com is 3599 sec (Scheduling reregistra tion in 3584 s) |
15:36.12 | a1fa | that is on sip reload |
15:36.23 | [TK]D-Fender | a1fa : news to me... |
15:36.26 | a1fa | ok |
15:36.29 | a1fa | it picked up this time |
15:36.33 | a1fa | after i did sip reload |
15:37.22 | a1fa | canreinvite=no |
15:37.28 | a1fa | right? |
15:37.42 | salviadud | you guys think linking vonage to asterisk is a pain, or a breeze? |
15:38.22 | [TK]D-Fender | a1fa : re-invites = evil |
15:38.31 | syle | i know vonage is scared of AT&T |
15:38.42 | syle | they dropped their rates 5 bucks after they did heh |
15:38.55 | cj-rm | How many people can asterisk realistically support in a single conference call? |
15:39.01 | [TK]D-Fender | salviadud :well you can't do it direct, you'd need an FXO channel for it... so "pain" it is |
15:39.12 | shiznatix | is there any reason why when i make a call through a zapata device to a regular phone line that my calls would go to a random phone number about 60% of the time? |
15:39.29 | key2 | how comes my x100p can Dial but can't detect when the line is ringing, any idea ? |
15:39.38 | Katty | it's deaf. |
15:39.39 | *** join/#asterisk Letron (n=asd@216.94.46.194) |
15:39.47 | [TK]D-Fender | Katty : lol |
15:40.07 | pdunkel | shiznatix: That shoul make you popular in you neighborhood. I hope you don't develop during night-time! |
15:40.40 | Katty | [TK]D-Fender: lolerskater. |
15:40.43 | *** join/#asterisk Ansonmus (n=ahaeser@dsl97-13-100.fastxdsl.nl) |
15:40.56 | cj-rm | [TK]D-Fender: Did you find out what was up with my dialplan for channel groups? I'm sorry but I got dragged away by some other work... |
15:40.58 | Ansonmus | hello, do anyone have experioence with Express Talk? |
15:41.01 | shiznatix | pdunkel, hahahahaha. so far i have gotten 3 different people and im like uhhhh sorry my PBX is sending out random singlas.... |
15:41.23 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
15:41.23 | [TK]D-Fender | key2 : Do you have a context defined in zapata.conf and properly matching in extensions.conf for that channel? |
15:41.56 | pdunkel | shiznatix: What channels are you using? |
15:42.21 | *** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-029.mycingular.net) |
15:42.38 | stoffell | oh boy, Ukraine does not "certify" asterisk/digium equipment ??! |
15:42.42 | *** part/#asterisk justinu|laptop (n=Justin@wirelessdata-031-029.mycingular.net) |
15:42.55 | a1fa | * > Ukraine |
15:42.57 | a1fa | lol |
15:43.12 | pdunkel | stoffel: ? Where's that from? Ukraine? So? |
15:43.24 | salviadud | are there sexy girls in ukraine? |
15:43.35 | *** join/#asterisk AlexCTI (n=alex@68-66-149-78.miamfl.adelphia.net) |
15:43.54 | stoffell | pdunkel, a company doesn't want to use asterisk because they don't have a certificate for digium equipment on the telco's in ukraine |
15:43.58 | pdunkel | salvidud: Yes there are. I guess I'll have to video call them via an * in Europe then :) |
15:44.41 | Katty | there are pretty girls everywhere. |
15:44.44 | [TK]D-Fender | cj-rm : Really not sure... |
15:45.28 | pdunkel | stoffel: Well, I guess that makes that company not very smart. I also guess the don't want to use */digium for other reasons and use Ukraine as an excuse. Sort of like me saying I'm not using asterisk because it's not ceritfied by (me). |
15:46.22 | cj-rm | [TK]D-Fender: It's really strange, it looks like when I Dial(Zap/g1) that it somehow triggers the same internal extension that is running. |
15:46.28 | pdunkel | shiznatix: Still want me to try and help with your dialout problem? |
15:46.34 | zoa | like anyone in the ukraine cares about certificates |
15:46.44 | pdunkel | shiznatix: What channels? |
15:46.58 | key2 | pdunkel: well even if I didn't have, I would see something in the asterisk console ? |
15:47.10 | stoffell | zoa, that customer does :( |
15:47.11 | salviadud | well, there are pretty girls here, yet, i don't like their style |
15:47.21 | salviadud | i want me a black gurl! |
15:47.25 | pdunkel | key2? |
15:47.30 | salviadud | or something exotic, from the amazon |
15:47.40 | salviadud | a girl that likes to talk on the phone |
15:47.50 | DoktorGreg | What does the sendText command do? |
15:47.51 | salviadud | and have phone with my wacky ivr systems |
15:47.57 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
15:47.59 | *** join/#asterisk amonroy (n=chatzill@wireless-121.media.mit.edu) |
15:48.04 | salviadud | i meant, have fun |
15:48.10 | salviadud | ... damn, i'm sleepy |
15:48.12 | key2 | pdunkel: well if I ring the line, I would see something in the asterisk console no ? |
15:48.27 | pdunkel | saliva dud: mabe you should try the real thing some time. It is much more fun than on the phone! |
15:48.44 | DoktorGreg | will sendtext work with ISDN phones? |
15:48.46 | pdunkel | key2: That depends on verbosity. |
15:49.14 | *** join/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net) |
15:49.32 | [TK]D-Fender | key2 : pastebin your zapata.conf and extensions.con |
15:49.42 | pdunkel | key2 == shiznatix ? |
15:50.06 | salviadud | pdunkel, of course i've tried the real thing |
15:50.13 | salviadud | last night, i was doctor gonzo |
15:50.13 | *** join/#asterisk lecter___ (n=lecter__@200.218.192.10) |
15:50.16 | *** join/#asterisk leonk24 (n=adf@200.62.141.190) |
15:50.38 | pdunkel | saliva dud: You sure? And you still want phones**? |
15:50.51 | shiznatix | pdunkel, sorry, hold on let me try something real quick |
15:50.59 | leonk24 | Hi... somebody from Peru? I have a problem with circuit Telmex and TE110P |
15:51.00 | salviadud | hey, i like to talk on the phone too |
15:51.13 | lecter___ | Hi fellows. I'd like to know if ASTERISK can listen SIP messages at 5060 TCP. Is it possible? Or just UDP? |
15:51.24 | zoa | only udp |
15:51.27 | zoa | NEXT! |
15:52.31 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
15:52.54 | rollot | <PROTECTED> |
15:53.15 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
15:53.33 | lecter___ | second question: When I select TCP transport at a SIP device (I tried with Eyebeam 1.1), it just insert the field 'transport=TCP' in the headers but it sends using UDP. Is it right? |
15:54.05 | rollot | <PROTECTED> |
15:55.10 | noky | can i avoid to proxy rtp of asterisk ? ... i like that rtp will be between sipA & sipB directly... could be or asterisk have a problem with this or isn't implement of this form ? |
15:55.39 | ctooley | exten => _NXXNXXNXXX,1,Set,CDR(accountcode="14 - 2352"). I thought that was right, but it's not... what am I doing wrong? |
15:56.54 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
15:57.14 | *** part/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
15:58.09 | lecter___ | there are a ',' after set |
15:58.17 | [TK]D-Fender | ctooley : exten => _NXXNXXNXXX,1,Set(CDR(accountcode)=14 - 2352) |
15:58.31 | ctooley | [TK]D-Fender, finally found the wiki page that says that. |
15:58.33 | ctooley | thanks |
15:59.33 | leonk24 | Hi ... help me please with configuration for E1 Peru with Telmex. |
16:00.18 | leonk24 | I configured span with 1,1,0,ccs,hdb3,crc4 and load modules in the kernel but the led is red. |
16:00.55 | noky | is not possible that ast* disable the forward rtp between SIP_A & SIP_B ?... else ast* must control the signalling between SIP_A & SIP_B ? |
16:02.42 | shiznatix | OK! I have finally started getting much closer to getting faxes to work but with spandsp it is not really working |
16:03.10 | shiznatix | Asterisk understands its a fax, starts saving the fax, but then says that it was complete but there is no file saved anywhere |
16:03.40 | X-Rob | pdunkel, end result is, I left the code in there, commented out, and went 'This will be fixed when asterisk 1.4 is released' as a comment. Also hinted that TryExec will be the trick. Thanks 8) |
16:04.00 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
16:04.13 | noky | "The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. " |
16:04.14 | cybergypsy | i have an X100 card working for incoming calls , how do i dial( using it ? |
16:04.29 | noky | oh nice... where can i configurate that rtp may go between phones ? |
16:04.32 | backblue | cybergypsy: dial (Zap/1) ? |
16:04.33 | salviadud | asterisk 1.4? |
16:05.07 | salviadud | X-Rob, you workin' pretty close with the main Devs on the project? |
16:06.11 | noky | what should I do to make audio channels to go directly from phone to phone |
16:06.21 | cybergypsy | thanks backble - i tried that and it gave me an error , it seemed to obvious |
16:06.54 | cj-rm | What does /n do to the Local channel??? |
16:06.58 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
16:07.59 | *** part/#asterisk hensema (n=erik@scrat.hensema.net) |
16:08.03 | rollot | <PROTECTED> |
16:11.57 | cj-rm | Does the local channel work properly with Zap channel groups? |
16:12.01 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
16:15.17 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net) |
16:15.30 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
16:15.45 | *** join/#asterisk mjackson (n=mjackson@69.85.202.2) |
16:15.54 | mjackson | howdee howdee |
16:16.57 | salviadud | howdee partner |
16:16.59 | mjackson | Thank you for calling #asterisk. Your call is very important to us. Please stay on the line. Your call may be recorded for quality control purposes. |
16:17.25 | [TK]D-Fender | mjackson : No, probably more for future public ridicule ;) |
16:17.29 | salviadud | damn, i a called a friend at work |
16:17.44 | salviadud | i was all uppity, and the guy was like "dude.... i'm at work" |
16:18.03 | salviadud | man, i felt the coldness |
16:18.38 | salviadud | where i work, it's madness, so i can fun at work... i've never really had a serious job in my life |
16:18.56 | salviadud | i mean, have fun |
16:19.07 | mjackson | You at work now? :-) |
16:19.12 | salviadud | yeah :) |
16:19.30 | mjackson | ^^ |
16:19.30 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
16:19.36 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
16:19.41 | mjackson | Anybody got a preference between gnudialer and vididial? |
16:20.11 | mitcheloc | mjackson: can you link me to vididial? |
16:20.31 | mjackson | I just got done using festival to set up a new submenu in the menu system... it's I wonder if anybody will notice the 'accidental' studdering the robot voice does |
16:20.43 | salviadud | i'm like "hey mr. boss man, i'm gonna get us a toll free number service so we can call those hotlines" he'll go something like this "COUGH!!!@@1!umm... yeah, whatever, you do what you gotta do" |
16:20.46 | *** part/#asterisk leonk24 (n=adf@200.62.141.190) |
16:20.54 | *** join/#asterisk leonk24 (n=adf@200.62.141.190) |
16:21.12 | shiznatix | Alright im back! My Zap device is dialling a wrong number about 40% of the time can anyone help me out? |
16:21.34 | salviadud | some dude was playing with festival the other day. he made it go "domo origato, mr. roboto" |
16:21.50 | salviadud | it sounds kinda funky |
16:21.55 | mjackson | LoL |
16:23.26 | mjackson | ack... he says it all wrong... would take some tweaking |
16:23.53 | salviadud | yeah, nothing like fine tuning the instruments we use |
16:23.54 | austinnichols102 | festival is excellent for the stephen hawking voice |
16:23.58 | austinnichols102 | "Einstein was very unhappy about this apparent randomness in nature. His views were summed up in his famous phrase, 'God does not play dice'. " |
16:24.00 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
16:24.30 | cj-rm | Has anyone here looked at getting Nuances ( http://www.nuance.com/ ) excellent tts voices working with asterisk? |
16:24.32 | mjackson | mchawking.com |
16:24.39 | mjackson | hawking's voice makes great rap music |
16:24.43 | austinnichols102 | cepstral diane is excellent |
16:24.44 | cj-rm | They've got a good interactive demos on their site. |
16:25.23 | cj-rm | The British Male voice is particularly good... very much the BBC newscaster. |
16:26.24 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
16:26.29 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
16:27.12 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
16:27.34 | austinnichols102 | mjackson: nice link for mchawking.com |
16:27.59 | [TK]D-Fender | Cepstral seems an affordable TTS solution... |
16:28.02 | salviadud | i'm checking the faq out |
16:28.07 | salviadud | very funny stuff |
16:28.09 | [TK]D-Fender | In as much as I'd like "free" |
16:29.07 | *** part/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net) |
16:32.08 | mjackson | Anybody got some advice for somebody about to set up their first predictive dialer center? 40 people to start, growing to 200? |
16:35.21 | xachen | with Asterisk? |
16:35.29 | mjackson | yah |
16:35.33 | mjackson | i think we're going to use gnudialer |
16:35.37 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
16:38.54 | mjackson | gnudialer is an asterisk addon :-P |
16:40.56 | docelm0 | mjackson, check out vicidial.. |
16:41.02 | docelm0 | I know the author.. Good guy |
16:41.22 | op3r | mjackson: vicidial is cool |
16:41.51 | docelm0 | He lives about 20 miles from me |
16:42.32 | cj-rm | mjackson: My advice is dont... The world could use fewer cold calls. |
16:43.10 | docelm0 | cj-rm, true.. but gotta make money somehow |
16:43.25 | op3r | hahahahaha |
16:44.10 | cj-rm | docelm0: So do drug dealers, pimps and the mafia... |
16:44.32 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk Ansonmus (n=ahaeser@dsl97-13-100.fastxdsl.nl) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk tessier (n=treed@69.43.173.37) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk Supaplex (n=supaplex@shell.aros.net) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk De_Mon (n=de_mon@fl-67-77-160-188.dyn.sprint-hsd.net) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk Abydos313 (i=justcall@72.20.3.66) [NETSPLIT VICTIM] |
16:44.32 | *** join/#asterisk RaYmAn-Bx (i=rayman@80.163.27.65) [NETSPLIT VICTIM] |
16:45.37 | Hmmhesays | anyone familiar with MS's dns server? |
16:46.05 | Hmmhesays | yeah, i have no choice in this matter |
16:46.14 | Hmmhesays | so save your sarcasm |
16:46.32 | mjackson | lol |
16:46.34 | salviadud | lol |
16:46.55 | Hmmhesays | yes we're all comics |
16:46.59 | xachen | call centers... peh :( |
16:47.24 | xachen | I don't care if you want to sell me a $2000 Mr. Cosmo vacuum when I can get a just as good one from Sears for $200 :P |
16:47.54 | sevard | xachen: telemarket trap script |
16:47.55 | mitcheloc | Hmmhesays: use dnsmadeeasy ;) |
16:48.10 | xachen | heh |
16:48.27 | xachen | yeah when I get my new system up it'll be knocking off telemarketers |
16:48.51 | sevard | <PROTECTED> |
16:49.12 | Hmmhesays | i just have a simple question on how ms sets up their A records, because I can ping this www.foo.bar but not foo.bar |
16:49.49 | Hmmhesays | in bind this is simple, in ms... not so much |
16:49.52 | docelm0 | Hmmhesays, ya |
16:49.57 | docelm0 | What do you wanna know? |
16:50.04 | docelm0 | I use it for my primary DNS |
16:50.14 | Hmmhesays | ms? |
16:50.18 | docelm0 | yep |
16:50.20 | docelm0 | I use active directory |
16:50.28 | docelm0 | No choice.. Was in place when I got here |
16:50.29 | Hmmhesays | how do I make foo.bar return the ip |
16:50.34 | docelm0 | @ |
16:50.58 | docelm0 | or setup an A record with foo.bar. point to the IP |
16:51.08 | NirS | any digium wizards around here ? |
16:51.08 | docelm0 | make sure you use the . at the end of the same |
16:51.08 | Hmmhesays | won't let me add it in the mmc |
16:51.50 | docelm0 | Your wanting to resolve foo.bar to an ip.. Under Zone and then the name tell it you want to add a new A record with foo.bar. as the name and your IP |
16:52.01 | Hmmhesays | oohhh gotcha |
16:52.03 | iCEBrkr | *** glibc detected *** corrupted double-linked list: 0x08259650 *** |
16:52.03 | iCEBrkr | Aborted |
16:52.14 | docelm0 | haha |
16:52.37 | iCEBrkr | Find.. I'll build 1.2.6 |
16:52.38 | iCEBrkr | geesh |
16:52.39 | xachen | haha |
16:52.48 | xachen | asterisk is evil to compile |
16:52.53 | docelm0 | #0 0x008b5911 in ____strtod_l_internal () from /lib/tls/libc.so.6 |
16:52.53 | docelm0 | #1 0x008b2f68 in __strtod_internal () from /lib/tls/libc.so.6 |
16:52.53 | docelm0 | #2 0x008aefcd in atof () from /lib/tls/libc.so.6 |
16:52.53 | docelm0 | #3 0x006f1f43 in ?? () from /usr/lib/asterisk/modules/cdr_addon_mysql.so |
16:52.57 | docelm0 | explain that one |
16:52.58 | docelm0 | :P |
16:53.23 | Hmmhesays | wtf, the new host section of the mmc won't let me enter a . |
16:53.38 | docelm0 | Hmmhesays, Im a 2000 and 2003 MCSE |
16:53.54 | docelm0 | You cant enter a . you enter: foo.bar. |
16:53.59 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
16:54.32 | Hmmhesays | foo.bar. yes it comes out foobar |
16:54.33 | docelm0 | atof.. shit.. I know where the problem is.. DAMNIT! |
16:54.51 | docelm0 | um, hmmm |
16:55.03 | docelm0 | hold on lemme get into my DNS |
16:56.05 | docelm0 | ok load up the DNS manager.. Then click on Forward Lookup Zones then the domain name. Once you click on the name |
16:56.42 | Hmmhesays | is this something different than what i'm using? i just got the dns mmc up |
16:57.24 | docelm0 | no.. DNS MMC |
16:57.31 | Hmmhesays | i got a window with the name/mail/sub domains.. |
16:57.33 | docelm0 | Tell it to add a new host.. A record |
16:57.44 | docelm0 | What version windows are you using? |
16:57.46 | Hmmhesays | right click, new host |
16:58.03 | Hmmhesays | 2000 server sp4 |
16:58.22 | docelm0 | When you add the new host leave "NAME" bank the first field.. 2nd will be greyed out and enter your IP in the 3rd.. |
16:58.34 | docelm0 | ya.. same MMC plugin then |
16:59.01 | mjackson | you don't have to use the mmc |
16:59.09 | mjackson | i think you can directly edit your zone files on an ms dns server |
16:59.14 | mjackson | and they're standard format zone files |
16:59.19 | docelm0 | NOPE! |
16:59.25 | docelm0 | You have to use the GUI |
16:59.32 | mjackson | LoL |
16:59.48 | docelm0 | And they are not standard format.. M$HIT once again has their own way to do it. |
17:00.05 | Hmmhesays | yeah that worked docelm0, thanks a bunch |
17:00.15 | docelm0 | No prob dude.. |
17:00.42 | docelm0 | What can I say Im a well rounded engineer.. Sun, Linux, WinDOZ, networking, cisco, asterisk, etc.. I could go on FOREVER |
17:00.48 | mjackson | http://en.wikipedia.org/wiki/Microsoft_DNS DNS Data can be stored in zone files, or in Active Directory. MS DNS can be administrated via a GUI, the MMC, or a command line interface dnscmd |
17:01.20 | docelm0 | mjackson, read the last thing is says.. dnscmd from the CLI |
17:01.27 | docelm0 | you cant not directly edit the files |
17:01.30 | mjackson | *nod* |
17:01.31 | docelm0 | its a BAD ideas |
17:01.32 | sevard | docelm0: I want to hear you go on forever. |
17:01.33 | docelm0 | err idea |
17:01.36 | mjackson | right... if it's stored in active directory |
17:01.57 | docelm0 | I never store my in AD.. too much headache |
17:02.00 | mjackson | i bet each method handles editing the information differently... and each has it's own quirks. Yuck. |
17:02.02 | docelm0 | anywho.. going to lunch. |
17:02.23 | mjackson | yay lunchtime! |
17:03.08 | *** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net) |
17:04.39 | iCEBrkr | docelm0: you're well 'rounded' alright. |
17:04.43 | sevard | i ate my lunch at 11 :/ |
17:05.42 | timscott | it's 11 right now. :/ |
17:05.45 | timscott | MST |
17:05.46 | timscott | :/ |
17:06.39 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
17:09.08 | Hmmhesays | i need a cigarette |
17:09.23 | *** join/#asterisk Nix (n=Nix@81.214.255.57) |
17:18.18 | zaf | AD-integrated DNS doesn't work too bad |
17:23.21 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
17:26.37 | Hmmhesays | wow trying to troubleshoot ser with techs that don't know sh1@t it rough |
17:31.23 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
17:31.39 | timscott | ser? |
17:32.03 | timscott | looks like the techs aren't the only ones who don't know anything about it... |
17:32.04 | timscott | :S |
17:32.43 | *** join/#asterisk gmonxx (n=gg@65.172.4.34) |
17:33.24 | Qwell[] | timscott: What's wrong with SER? |
17:34.13 | gmonxx | anyone know why my voice volume is really low when making an outside call? |
17:35.19 | timscott | Qwell: I don't know. I don't know _what_it_is_. |
17:35.21 | timscott | :/ |
17:37.02 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
17:37.07 | [av]bani | ~ser |
17:37.13 | jbot | methinks ser is Sip Express Router - see http://www.iptel.org/ser/ |
17:38.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:40.50 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
17:41.58 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
17:44.35 | a1fa | i need to pee |
17:44.46 | Qwell[] | good to know |
17:44.52 | *** join/#asterisk Dandan (i=dandan@jestem.lama.ale.mam.super.konto.na.pacanka.com) |
17:44.54 | Dandan | hey all |
17:45.50 | a1fa | thankz qwell |
17:45.53 | docelm0 | Hay TIM uhh screw you there bub |
17:46.06 | Dandan | i have a question: I will be getting a PRI, what card would you recommend? |
17:46.16 | Dandan | (I decided to drop 15 copper lines...) |
17:46.30 | Qwell[] | Dandan: a single PRI? Digium TE110P or TE105P |
17:46.50 | mog_work | we have no 105........... |
17:46.51 | Dandan | Qwell: cool, no problems whatsoever? |
17:46.56 | Qwell[] | oh |
17:47.00 | Dandan | i decided to drop voicetronix |
17:47.09 | Qwell[] | well, I just made that one up then :p |
17:47.10 | Dandan | couldn't get it to work one way or another... |
17:47.11 | [av]bani | yay software EC |
17:47.25 | Qwell[] | mog_work: Do you only have one voltage for the 1 port? |
17:47.38 | mog_work | rigth it does 3.3 or 5 for that board |
17:47.47 | Qwell[] | ahh, okay |
17:47.48 | Dandan | anyone heard anything good/bad about CTC Comm as providers? |
17:48.04 | Qwell[] | So it's the TE110P |
17:48.15 | Qwell[] | mog_work: Why don't the other boards do both? |
17:48.46 | *** join/#asterisk jeffik (n=Jeff@208-41-192-106.client.dsl.net) |
17:48.57 | mog_work | xylinx thing |
17:49.01 | Qwell[] | oh |
17:49.07 | Qwell[] | silly xylinx :D |
17:49.10 | wunderkin | :( |
17:49.12 | mog_work | man i just butchered that word |
17:49.21 | mog_work | i think its xilinx but i have no clue |
17:49.41 | Qwell[] | yeah, it is |
17:49.51 | Qwell[] | per google |
17:50.12 | a1fa | brb |
17:50.14 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
17:51.58 | jeffik | Anybody know how to access sipura 1001 remotely? |
17:52.04 | Dandan | anyone has any experience with sangoma a101 |
17:52.06 | Dandan | ? |
17:52.26 | brif8 | I have bought a cisco 7920 but they say without the CM license where can I just buy the CM license so I can get the cmterm file anyone >? |
17:52.51 | *** join/#asterisk maffro (n=furby@n156.dkm.cz) |
17:52.59 | maffro | Hi all |
17:53.18 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
17:53.24 | a1fa | ls |
17:53.24 | *** join/#asterisk oconnect (i=lukash@sip.pekelnik.net) |
17:53.26 | a1fa | sudo su- |
17:53.30 | a1fa | r0#*@($#*$#R#RJ |
17:53.36 | a1fa | omfg ;) |
17:53.36 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
17:53.41 | a1fa | you have my rewt password |
17:54.03 | [TK]D-Fender | Dandan : All the Sangoma cards work the same. I have an A104d |
17:54.20 | a1fa | [TK]D-Fender : can you make a decent living doing voip consulting? |
17:54.24 | Qwell[] | brif8: Any cisco reseller should be able to sell you a CM license...not that you need it |
17:54.24 | Qwell[] | You do need a smartnet contract though |
17:54.30 | [TK]D-Fender | jeffik : Tried using a web-browser? |
17:54.36 | Qwell[] | a1fa: You can, sure |
17:54.46 | a1fa | Qwell[] : what planet? |
17:54.47 | Dandan | [TK]D-Fender: hey :) |
17:54.50 | a1fa | or city? |
17:54.55 | Dandan | i decided to return the voicetronix |
17:55.08 | techman97_andy | hello all again - my system accepts calls from a SIP provider, and I'd like there to be 1-2 rings before Asterisk picks up and dumps into my IVR. I currently have a "wait(2)" in my extensions.conf file, but that's just dead noise. Any idea on what commands I can use to provide ringtone to the caller? |
17:55.10 | Dandan | i had SO many issues that it was rediculous |
17:55.14 | Qwell[] | a1fa: There are quite a few people who do it |
17:55.16 | Dandan | went with CTC Comm and PRI |
17:55.22 | Dandan | so now I need a new card |
17:55.28 | a1fa | techman97_andy : yes |
17:55.31 | [TK]D-Fender | Dandan :REALLY?! ;) |
17:55.38 | Dandan | and I came here for an advice |
17:55.44 | a1fa | exten => s,2,Playtones(ring) |
17:55.47 | [TK]D-Fender | Looking for PRI this time are we? |
17:55.52 | techman97_andy | a1fa: cool - thanks! |
17:55.52 | brif8 | Qwell[]: How and where I've tried calls to cisco atacomm ingram micro I just want the cmt bin files |
17:55.55 | a1fa | ok |
17:56.02 | a1fa | stupid * is not picking up a call again |
17:56.05 | a1fa | wtf |
17:56.21 | ruza | lukash: ':) |
17:56.26 | lukash | ruza: :) |
17:56.27 | a1fa | ok |
17:56.29 | a1fa | wtffff |
17:56.30 | a1fa | i am pissed off |
17:56.34 | *** join/#asterisk noky (n=Noky@200.69.211.18) |
17:56.35 | noky | hi |
17:57.17 | brif8 | Qwell[]: can you recommend a reseller who can sell it to me now on line ? |
17:57.26 | Dandan | fenlander: oh, shshsh... |
17:57.26 | Dandan | [TK]D-Fender: yes :/ |
17:57.26 | Dandan | do not say "didn't I say so..." |
17:57.26 | Dandan | 15 lines are a bit too many to interface with voicetronix... |
17:57.30 | Dandan | [TK]D-Fender: so there are no issues with IRQs/drivers under linux 2.6? |
17:57.48 | docelm0 | MEW MEW MEW MEW MEW MEW MEW MEW MEW |
17:57.54 | Qwell[] | techman97_andy: Wait() should work...just don't Answer() beforehand |
17:57.55 | a1fa | broadvoice's sip proxies are as reliable as your friendly neighborhood drug dealer |
17:57.56 | [TK]D-Fender | Dandan : With what? |
17:58.02 | Dandan | sangoma |
17:58.14 | [TK]D-Fender | Dandan : Not one case I can recount. |
17:58.16 | syle | not reliable then hehe |
17:58.22 | Dandan | a1fa: I had no problems with BV today... |
17:58.31 | Dandan | it was really bad friday but not today |
17:58.33 | a1fa | Dandan : what proxy? |
17:58.35 | Qwell[] | brif8: look on the wiki, for cisco smartnet |
17:58.43 | noky | "The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. " |
17:58.45 | Dandan | hm, do not recall, hold on :D |
17:58.46 | Qwell[] | it'll list a reseller or two |
17:58.49 | jeffik | D-Fender: yes, using extenal ip and set the port to 8022 |
17:58.54 | noky | what do u trhink about this [TK]D-Fender ? |
17:58.58 | noky | think* |
17:59.00 | [TK]D-Fender | Dandan : Let me just say that unless your company is UBER-cheap (aka STUPID), get the A104d.. the hardware EC is entirely worth it. |
17:59.02 | Dandan | [TK]D-Fender: ty going to tell my purchasing dept to buy it for me |
17:59.22 | a1fa | Apr 11 17:59:10 NOTICE[11439]: chan_sip.c:9686 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) |
17:59.25 | a1fa | i dont get it |
17:59.26 | [TK]D-Fender | noky : Re-invites are EVIL. <- |
17:59.37 | a1fa | why is it re-registering every 30s |
17:59.42 | a1fa | i told it to re-register every hour |
17:59.44 | Dandan | [TK]D-Fender: quad pris? |
17:59.44 | a1fa | god damn it |
17:59.46 | techman97_andy | a1fa: that did it - thanks again! |
17:59.57 | a1fa | yeah, you are welcome |
18:00.00 | Hmmhesays | IIS is going to make me rip my eyeballs out |
18:00.08 | [TK]D-Fender | Dandan : Yes, I know you don't need 4 ports, but thats the only size out now with the HWEC on it. |
18:00.08 | a1fa | guys |
18:00.13 | a1fa | heeeeelp |
18:00.20 | [av]bani | wait 12mo for sangoma to make a 2port ver |
18:00.22 | Dandan | hm |
18:00.23 | a1fa | Outbound Registration: Expiry for sip.broadvoice.com is 30 sec |
18:00.24 | [av]bani | simple |
18:00.26 | a1fa | its not |
18:00.31 | Dandan | well, voicetronix was supposed to have EC... |
18:00.37 | Dandan | and all I was getting was ECHO :D |
18:00.41 | [TK]D-Fender | [av]bani : Should be sooner than that.... |
18:00.45 | noky | mmm... i want to know if is posibble that Asterisk could not forward rtp... |
18:00.51 | a1fa | maxexpirey=180 |
18:00.51 | a1fa | defaultexpirey=160 |
18:00.52 | [av]bani | or buy an external sip gateway |
18:01.02 | Dandan | a1fa: 147.135.20.128 sip.broadvoice.com |
18:01.07 | a1fa | nyc |
18:01.09 | a1fa | i got the same one |
18:01.16 | noky | i want to my rtp goes between Phone A & Phone B... not Phone A => Asterisk => Phone B |
18:01.18 | Dandan | on choiceone as a provider... |
18:01.32 | noky | possible** |
18:01.36 | a1fa | ok |
18:01.37 | a1fa | wtf |
18:01.42 | a1fa | why is it re-registering every 30s |
18:01.48 | a1fa | when it is told to re-register every fucking hour |
18:01.53 | [TK]D-Fender | noky : If you use "canreinvite=yes" on both sides of the call then they will reconnect directly to each other (usually a BAD idea). So I suggest "canreinvite=no" as a global setting |
18:02.00 | DoktorGreg | oh man |
18:02.11 | DoktorGreg | sorry ive been deeply in identity crisis |
18:02.24 | noky | ok thanks [TK]D-Fender ! |
18:02.41 | DoktorGreg | But apple has release a rdc that lets you drag and drop accross computers... |
18:02.46 | [av]bani | why is reinvite bad, unless you're in nat? |
18:02.57 | a1fa | does anybody know why am I getting this stupid shit |
18:02.59 | [av]bani | reinvite is the only way to scale to 1000's of extensions on a single box |
18:03.04 | a1fa | Outbound Registration: Expiry for sip.broadvoice.com is 30 |
18:03.20 | Hmmhesays | turn off your monitor |
18:03.31 | a1fa | ? |
18:03.34 | Dandan | [TK]D-Fender: thx, i will look into getting that 104d |
18:03.34 | a1fa | Hmmhesays : lol |
18:03.45 | sevard | where are you tftpd |
18:04.11 | [av]bani | Dandan: the te1** cards are also very finicky about the motherboards they will work in, the 104d is guaranteed to work in anything |
18:04.22 | [TK]D-Fender | [av]bani : because typically phones don't end up getting public IP's. I guess you could isolate your outside peers, but better to just let it flow through unless you're huge |
18:04.29 | a1fa | [TK]D-Fender : yo yo yo |
18:04.35 | Dandan | [av]bani: i have a dell sc430 |
18:04.37 | [av]bani | [TK]D-Fender: if your extensions arent talking to the outside world... |
18:04.39 | Dandan | as my test platform |
18:04.58 | [av]bani | [TK]D-Fender: which is the typical case... your extensions always talk through the pbx, not directly to the intarweb |
18:04.59 | [TK]D-Fender | [av]bani : they do sometimes. |
18:05.20 | [av]bani | [TK]D-Fender: i would say 99% of the time that is not hte case... 99% of the time people are using * for internal pbx, not public intarweb pbx |
18:05.21 | Hmmhesays | if you don't have a massive number of calls, who cares |
18:05.35 | grem_lin | Hey, could anyone comment on the suitability of the Zyxel Prestige 2002 for use in conjunction (or, indeed general use) with asterisk before I go and buy one ? :) |
18:05.36 | [TK]D-Fender | Hmmhesays : My thoughts exactly... |
18:05.48 | [TK]D-Fender | grem_lin : Bleh <- |
18:05.56 | [TK]D-Fender | Wifi + SIP = shit today. |
18:05.59 | noky | [TK]D-Fender: thanks. |
18:06.00 | [av]bani | Hmmhesays: if you're running on low end hardware (wrt54gs, gumstix), then you want reinvite |
18:06.08 | a1fa | wifi + iax = future |
18:06.13 | a1fa | [TK]D-Fender : hey fender |
18:06.14 | [av]bani | Hmmhesays: of course, you could wimp out and say "dont run asterisk on low end hardware", which is lame |
18:06.32 | [TK]D-Fender | Dandan : the SC430 is on Digium's "shit list" IIRC with regards to several of their cards. |
18:06.33 | [av]bani | Hmmhesays: think outside the box. it must get cramped in there. |
18:06.42 | a1fa | gumstix * > any |
18:06.54 | Dandan | [TK]D-Fender: sh*t, you think I should get a sangoma then? |
18:06.54 | [TK]D-Fender | iax + anything other than other IXA server = toy |
18:07.03 | a1fa | hehe |
18:07.03 | Dandan | (I have to present a working asterisk server by the end of the month) |
18:07.11 | grem_lin | Oh don't say that :( I've spent ages trying to find the right piece of kit |
18:07.16 | Dandan | otherwise avaya here i come... |
18:07.18 | a1fa | [TK]D-Fender : hey man.. i am having an issue with re-registration.. it is expriging every 30s |
18:07.30 | a1fa | [TK]D-Fender : expirying |
18:07.39 | a1fa | or howerver you spell expire + ing |
18:07.42 | [TK]D-Fender | Could be the OTHER side.... not you at all.. maybe they're assy today |
18:07.57 | a1fa | [TK]D-Fender : its been doing this since i updated to 1.2.6 |
18:08.12 | a1fa | [TK]D-Fender : and they keep telling me they dont have that setting set to 30s |
18:08.38 | a1fa | i know they do |
18:08.45 | Dandan | well, recently asterisk's pace of development outpaces kernel's. Amazing. TVG train should dash for cover... |
18:09.00 | a1fa | handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec |
18:09.31 | [TK]D-Fender | Dandan : If the telco isn't stupid you can set a small company up in a day from scratch. |
18:09.47 | [TK]D-Fender | Dandan : Pick good phones.... |
18:10.26 | a1fa | omfg |
18:10.31 | a1fa | [TK]D-Fender : you are right |
18:10.36 | a1fa | it is coming from broadvoice |
18:10.59 | Dandan | [TK]D-Fender: i just need a foolproof setup, i grandfathered the sc430, now I need a pri card for it... |
18:11.17 | [TK]D-Fender | rule of thumb : ITSP's suck. ALL of them. Just some more than others at varying intervals... |
18:11.50 | [TK]D-Fender | Dandan : the SC430 is your first mistake. PM your expected setup / needs / expectations. |
18:13.02 | a1fa | http://pastebin.ca/49047 |
18:13.04 | a1fa | [TK]D-Fender http://pastebin.ca/49047 |
18:13.07 | a1fa | SIP READ |
18:13.16 | a1fa | is config coming down from the server.. rgith? |
18:13.39 | a1fa | Expires: 30 |
18:13.44 | a1fa | god damn fucking bastards |
18:13.51 | a1fa | i fucking hate these motherfuckers |
18:14.03 | a1fa | i am not able to recieve a call forever |
18:14.07 | [TK]D-Fender | a1fa : since you changed the names I presume the "from" was THEM, and the ""to" was YOU? |
18:14.32 | a1fa | yes |
18:14.36 | maffro | 000000000000. |
18:14.40 | a1fa | From: was me |
18:14.42 | a1fa | brimstone: was me |
18:14.48 | a1fa | "To:" was me |
18:15.23 | [TK]D-Fender | :/ |
18:15.38 | [TK]D-Fender | wierd... not sure how to interpret that... |
18:15.51 | a1fa | :) |
18:16.06 | a1fa | its crazy man |
18:16.15 | a1fa | SIP READ FROM 147.135.20.128:5060: |
18:16.28 | a1fa | this means that i am getting data from them, right? |
18:16.51 | a1fa | "From: <sip:me@sip.broadvoice.com>" |
18:16.55 | a1fa | "To: <sip:me@sip.broadvoice.com>" |
18:17.04 | a1fa | "Expire: 30" |
18:17.08 | a1fa | god damn i hate these people |
18:17.11 | [TK]D-Fender | a1fa : I am thinking that'd be the case |
18:17.32 | [TK]D-Fender | you set "defaultexpiry" and all that for the peer? |
18:17.34 | a1fa | i am on hold |
18:17.46 | a1fa | defaultexpiry=3600 |
18:18.16 | [TK]D-Fender | a1fa : Global & peer? |
18:18.27 | a1fa | peer only |
18:18.35 | *** part/#asterisk Hali_303 (n=surfk@dsl51B6E6EB.pool.t-online.hu) |
18:18.58 | a1fa | global & peer now |
18:19.02 | a1fa | defaultexpiry=3600 |
18:19.10 | a1fa | i did sip reload |
18:19.11 | a1fa | firs message that came upo |
18:19.21 | a1fa | <PROTECTED> |
18:19.52 | Gamercjm | Does sox hangle ulaw? |
18:19.59 | Gamercjm | handle* |
18:20.14 | [TK]D-Fender | a1fa : Fuck the fucking fuckers ;) |
18:20.15 | Qwell[] | Gamercjm: yes |
18:20.17 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
18:20.37 | a1fa | [TK]D-Fender : those mother fuckers, i've been on hold for 30 minutes now |
18:20.46 | a1fa | god damn that fucking bastard and his call center |
18:20.47 | wunderkin | dont bother |
18:20.56 | pdunkel | Gamercjm: You might just as well type the words "sox" and "ulaw" into goolgle. Al the top posts tell you the answer. |
18:21.40 | maffro | hey folks, any of you enjoyed call recordings out of sync when using monitor to wav49 ? |
18:23.09 | Dandan | a1fa: nothing new |
18:23.14 | *** part/#asterisk Nix (n=Nix@81.214.255.57) |
18:23.26 | *** join/#asterisk Lino` (n=Lino@i577BCF08.versanet.de) |
18:23.28 | Gamercjm | Well i tried looking for it, but couldnt find the way its put into syntax "sox test.wav -r 8000 -c 1 -s -w test.ulaw resample -ql" |
18:23.32 | skyboy | hello has anyone used SER to load balance asterisk here? specifically what is the setup required to get it to go with srv records etc? |
18:23.34 | Gamercjm | does that look correct? |
18:23.47 | a1fa | ok |
18:23.50 | a1fa | i got those fuckers on the line |
18:24.24 | Beirdo | 976 number? |
18:26.27 | [hC] | [TK]D-Fender: any chance i could get that minibrowser stuff from you today? |
18:26.59 | a1fa | [TK]D-Fender : he sais he doesnt have that setting |
18:27.04 | a1fa | liar fucking bitch |
18:27.10 | a1fa | i am going to do a tcpdump |
18:27.16 | a1fa | and send him the binary log |
18:27.19 | a1fa | so he can use ettercap |
18:27.22 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
18:27.23 | a1fa | and jerk himself to death |
18:28.04 | pdunkel | Gamercjm: Almost there is no such extension .ulaw. the one used by sox to tell it is the pseudo extension .lu |
18:28.19 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
18:28.39 | pdunkel | Gamercjm: This information was just obtained for you by the command "man sox" any more quesstions and the next response is RTFM!!! |
18:29.00 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
18:29.04 | jpablo | people I have a big problem. I have a PRI when i dial a number that does not exists it gives a normal busy tone instead of playing the telco's "the number you are trying o dial does not exists, etc", how can i fix that ? |
18:29.43 | [TK]D-Fender | [hC] : sure. |
18:30.11 | [hC] | [TK]D-Fender: email work the best, or? |
18:30.14 | nettie | hey guys, I'm looking for a way to assign an extension number as CID for internal calls. I would like to have the extension displayed instead of the actual sip phone username. Anyone can suggestme something? I suppose I should use SetCallerID.. |
18:30.20 | jpablo | any idea how can i fix that ? |
18:30.40 | pdunkel | nettie: That might work :) |
18:30.49 | jpablo | the same pri works fine with a panasonic pbx |
18:31.11 | nettie | pdunkel yeah the problem is that if I user SetCallerID({$MACRO_EXTEN}) |
18:31.17 | nettie | s/user/use |
18:31.31 | nettie | it doesnt work |
18:31.43 | nettie | I get {$MACRO_EXTEN} displayed |
18:31.51 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
18:32.16 | pdunkel | nettie: Aren't there new CallerID functions? Hold I'll check. |
18:32.22 | kratzers | try ${MACRO_EXTEN} |
18:32.53 | kratzers | dollar sign then freedom brace |
18:33.05 | pdunkel | oej: Do you happen to know of hand? (If we have the honor of a guru present...) |
18:33.12 | Katty | [TK]D-Fender: have you seen clockwork orange? |
18:33.18 | nettie | and anyway I doubt it will help.. considering I would like to display the caller extension and not the called one |
18:33.20 | [TK]D-Fender | Katty : nope... |
18:33.25 | Katty | kk |
18:33.32 | nettie | kratzers I'm actually using curly |
18:33.33 | [TK]D-Fender | [hC] : PM me your e-mail addy |
18:33.37 | nettie | as you suggested |
18:33.39 | nettie | hey TK |
18:34.19 | pdunkel | nettie: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCIDNum details the new functions. |
18:34.56 | nettie | thanx a lot pdunkel checking... |
18:34.58 | Dandan | what is asterisk-addons-1.2.2-patch.gz ? |
18:35.05 | kratzers | say I have two agents logged in, and both go on do not disturb... is there a way for calls to be queued instead of goign to voicemail boxes? |
18:35.17 | pdunkel | nettie: That should evaluate the ${MACRO_EXTEN} expression since Set usually does that. |
18:35.33 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
18:35.38 | *** join/#asterisk Luda1 (n=nechci@rb1j38.chello.upc.cz) |
18:36.05 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
18:36.38 | Luda1 | hi, pls i need register my asterisk to voipcheap, can anybody help me ? |
18:37.01 | kratzers | it seems that agents who are on the phone or are on DND should not be considered eligible to take calls |
18:37.41 | malverian | I have a question about CDR... |
18:38.09 | *** join/#asterisk Dabian (n=M0RTEN@fsf/member/dabian) |
18:38.27 | oej | pdunkel: Know what? |
18:38.33 | nettie | pdunkel is does it.. the problem is that it sets the wrng parameters.. :( Executing Set("SIP/poly2-7260", "CALLERID(number)={$MACRO_EXTEN}") in new stack |
18:39.33 | Dandan | <PROTECTED> |
18:39.34 | kratzers | why is the curley brace before the dollar sign? |
18:41.34 | [TK]D-Fender | kratzers : to prevent it from working of course! |
18:41.44 | a1fa | [TK]D-Fender : lol.. they opened a ticket with broadvoice |
18:41.46 | a1fa | engineers |
18:41.47 | kratzers | ah |
18:41.51 | a1fa | stupid dump dumb fucks |
18:41.59 | a1fa | some fag changed expiry to 30 |
18:42.06 | a1fa | now they cant change it back to 3600 |
18:42.26 | kratzers | and by freedom brace in my previous answer, I meant the politically correct American version of french brace |
18:43.22 | mitcheloc | is anyone subscribed to the asterisk mailing list? i'm wondering if my message didn't go through |
18:43.37 | kratzers | I am |
18:43.38 | maffro | kratzers: heh, I for one was wondering |
18:44.07 | kratzers | maffro, it's a term my one Unix teacher used a few years back |
18:44.14 | mitcheloc | kratzers: could you check if a message from me went through (mitcheloc@gmail.com) |
18:44.31 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
18:44.40 | [TK]D-Fender | [14:42] <kratzers> and by freedom brace in my previous answer, I meant the politically correct American version of french brace <- since when is America "correct"? ;) |
18:45.08 | kratzers | mitcheloc: I don't see one |
18:45.09 | maffro | [TK]D-Fender: that comes after it is free |
18:46.08 | kratzers | so, why if there are two agents who are on calls and a new call comes in, is the new call sent to voicemail rather than queued? |
18:46.14 | mitcheloc | kratzers: hmm, is the list moderated? maybe they marked it as spam =/ |
18:46.16 | *** join/#asterisk Leob (n=chatzill@w2kvpn-22.media.mit.edu) |
18:46.22 | kratzers | doesn't that defeat the purpose of queueing? |
18:47.00 | kratzers | mitcheloc: I'm not sure, when did you send it? Maybe delivery is delayed? |
18:47.09 | mitcheloc | about an hour and a half ago |
18:47.31 | kratzers | hmm, try again maybe |
18:47.43 | [TK]D-Fender | kratzers : Using AgentCallbackLogin? |
18:47.57 | kratzers | [TX]D-Fender: yes |
18:48.04 | kratzers | bah |
18:48.15 | kratzers | stupid highlighting in irssi |
18:48.46 | kratzers | can't read your name because it is in yellow agains white |
18:48.52 | [TK]D-Fender | kratzers : then stop using a macro that answers the phone and dumps them to voicemail!~ |
18:49.02 | kratzers | hmm |
18:49.05 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
18:50.10 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
18:50.40 | Luda1 | hi, pls i need register my asterisk to voipcheap, can anybody help me |
18:50.45 | kratzers | I'm not sure that I am |
18:50.47 | [TK]D-Fender | What are you talking about? |
18:50.47 | kratzers | http://pastebin.com/654036 |
18:52.09 | [TK]D-Fender | kratzers : show us where agents loging and the context it dials into |
18:52.13 | PakiPenguin | [TK]D-Fender go the card link up atleast , but getting zaptel errors now |
18:52.38 | PakiPenguin | http://pastebin.com/653419 |
18:53.27 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
18:53.30 | [TK]D-Fender | PakiPenguin : the FXO stuff looks bad... |
18:53.59 | PakiPenguin | the fxo is the astribank i have |
18:54.00 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
18:54.16 | PakiPenguin | i mean fxo signaling |
18:54.19 | kratzers | relevant stuff -> http://pastebin.com/654046 |
18:54.22 | PakiPenguin | its fxs actuallly |
18:54.24 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
18:55.59 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
18:58.19 | PakiPenguin | [TK]D-Fender can you solve it ,o r point me where to solve it please |
18:59.25 | timscott | ERGGG |
18:59.26 | timscott | NAT |
18:59.30 | timscott | I hate NAT. |
18:59.33 | timscott | Seriously. |
19:01.51 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
19:01.57 | terrapen | man, the SPA942 ROCKZZZ |
19:02.52 | [hC] | I have a question actually, for anyone with the spa941/942... do you hear double-ringing with it? I dont hear double ring with my ciscos but i do with these phones. (and no im not specifying ,r to Dial) - also some people claim that these phones are a bit echoey |
19:03.04 | terrapen | double ringing? |
19:03.23 | [hC] | yeah, like you dial a number and rather than hearing one ring 'tone' |
19:03.26 | [hC] | you hear two, over lapped |
19:03.28 | terrapen | nope |
19:03.33 | [hC] | Hmm. |
19:03.41 | [hC] | I have it happen on two completely different systems. |
19:04.19 | terrapen | i'm doingprogressinband=no |
19:04.21 | terrapen | err |
19:04.21 | terrapen | progressinband=no |
19:04.28 | terrapen | try that |
19:04.41 | [hC] | i just read that in mantis |
19:04.42 | [hC] | i'll try it. |
19:04.51 | [hC] | im surprised it would be set to yes as default.. |
19:05.30 | [TK]D-Fender | PakiPenguin : Not sure about what to do with that... |
19:05.55 | [TK]D-Fender | [hC] : SPA-x4x = waste. |
19:06.03 | [hC] | hmm. that seems to have helped it. |
19:06.25 | Leob | hello there, can anyone help me with ODBC? Asterisk crashes all the time during VoiceMailMain... |
19:06.52 | [TK]D-Fender | kratzers : Well you are using a macro with an answer in there.. that screws your queues... you need to make another means of dialing them so they don't answer... |
19:07.29 | [TK]D-Fender | terrapen : Polycom IP501 can be had for less than the SPA-942 and is of superior usability & quality |
19:07.43 | [hC] | Yeah. |
19:07.49 | [hC] | I carry the 941/2 but i prefer the polycom |
19:08.05 | [hC] | if only i could get boot up time shortened, i'd be happy :) |
19:08.17 | [hC] | and this 7 line BLF resolved, we'd have a solid winner |
19:08.17 | kratzers | hmm, I suppose I need to look at a complete set of sample configs using agents to see how it should be done |
19:08.44 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
19:08.53 | [TK]D-Fender | kratzers : you did it in a "right way", just one with consequences... you need to make a context for your agents that only Dial's them, and nothing more... |
19:09.57 | [TK]D-Fender | [hC] : then again, how often do you need to reboot them? I did mine twice. Once to pick-up the config & firmware coming fresh out of the box, and the other to go live :) |
19:10.08 | kratzers | and that's the one they login to? |
19:10.17 | [hC] | [TK]D-Fender: power outages, etc.. |
19:10.20 | [hC] | for some clients |
19:10.30 | [TK]D-Fender | [hC] : UPS & PoE :) |
19:10.48 | terrapen | TDK, I'm not seeing how the polycom is more usable |
19:10.56 | terrapen | this has a backlit, very easy to read screen |
19:10.57 | [hC] | [TK]D-Fender: like i said.... 'some' customers. :) the cheap ones who dont vote for ups+poe |
19:11.01 | terrapen | sound quality is fine |
19:11.09 | terrapen | web interface is incredible, if you need it |
19:11.16 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
19:11.17 | [hC] | the 942 is backlit? |
19:11.22 | terrapen | yes |
19:11.24 | terrapen | white-lit |
19:11.28 | [hC] | i only have 941's |
19:11.50 | terrapen | oh, and the Linksys boots in less than 7 seconds |
19:11.56 | terrapen | compare to the Polycom's 2 minutes+ |
19:12.10 | brad_mssw | huh, the 942 is backlit ?? really? |
19:12.11 | terrapen | I've loved polycom but really, I think Linksys has them beat on this one |
19:12.14 | brad_mssw | crap, we just bought 941s |
19:12.16 | terrapen | brad, heck yeah, it's sweet |
19:12.23 | terrapen | its a white-lit LCD |
19:12.27 | Luda1 | hi, pls i need register my asterisk to voipcheap, can anybody help me ??? |
19:12.34 | terrapen | brad, but it's not as cool as my new mountain bike ;-) |
19:12.37 | [hC] | the 941 has a half-ass speakerphone too. |
19:12.45 | [hC] | the polycom beats it on speakerphone hands down |
19:12.49 | brad_mssw | terrapen: how is it that I didn't see that on the 942 spec sheet |
19:12.52 | h3x0r | the 941 is god awful |
19:13.03 | h3x0r | the sound quality is like a chinese phone |
19:13.04 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
19:13.05 | terrapen | hc, the speakerphone on the 942 is not the best evar but it's fine for desk use |
19:13.13 | terrapen | I may buy some 501's for the conference rooms |
19:13.16 | h3x0r | on the handset im talking about even |
19:13.25 | [TK]D-Fender | terrapen : Polycom uses the LCD more effectively, is larger and more readable under proper lighting. |
19:13.31 | [hC] | the sound quality could be better, for sure. |
19:13.33 | terrapen | the sound quality on the handset for the 942 is great |
19:13.43 | Shaun2222 | where can i find a list of valid tone's for a dialplan on a 7960 cisco phone? |
19:13.45 | [hC] | it sounds like they may have improved the phone alot w/ the 942 |
19:14.04 | [TK]D-Fender | h3x0r I wouldn't say the 941 is "bad", just not comparable to something similarly priced. |
19:14.06 | terrapen | TK, dunno, I have them side-by-side right in front of me. This is the 942 I'm talking about |
19:14.17 | terrapen | Besides, the polycom's minibrowser sucks |
19:14.20 | kratzers | think it's working now, but I'll have to play with it more tomorrow |
19:14.22 | kratzers | thanks |
19:14.29 | terrapen | TK, I'm starting to think that the 942 is much-improved over the 941 |
19:14.33 | [hC] | terrapen: the linksys doesnt even have a minibrowser. does it? |
19:14.40 | terrapen | yeah, I think it does. |
19:14.43 | terrapen | this one does. |
19:14.46 | [hC] | hm. |
19:15.03 | [TK]D-Fender | terrapen : But costing more than the 501 really kills it... and I doubt the display is any better utilized. backlight IS a nice bonus though.... |
19:15.04 | [hC] | the 941 has 'directory' |
19:15.04 | [hC] | but thats it |
19:15.08 | terrapen | well, maybe not. i'm not sure |
19:15.12 | [hC] | haha |
19:15.17 | [hC] | the 941 also has an option in the menu |
19:15.20 | [hC] | 'Call Foward' |
19:15.21 | [TK]D-Fender | No, Linksys does NOT have a microbrowser |
19:15.24 | terrapen | how much is the 501, tk |
19:15.49 | [TK]D-Fender | [hC] : Polys have forwarding as well, and can do it on a per-contact basis too |
19:15.59 | [TK]D-Fender | terrapen : Seen for jsut under $170USD |
19:16.10 | [hC] | I was just making fun of how they missed the r in forward. |
19:16.12 | [hC] | 'Foward' |
19:16.16 | brad_mssw | terrapen: dunno, dude, can't find anything about backlights on the 942 |
19:16.23 | terrapen | ok, no minibrowwser |
19:16.38 | terrapen | brad, i swear to you, it is. :) I will take a picture tonight |
19:16.48 | terrapen | it also helps greatly in bright light situations |
19:17.05 | brad_mssw | terrapen: yeah, i'd like to see it, these damn 941s can be hard to read |
19:17.47 | *** join/#asterisk justinu|laptop (n=Justin@66.209.15.236) |
19:17.59 | terrapen | i used this phone last night in the dark office and it was great |
19:18.17 | brad_mssw | they come with the 2-port built-in switch as well, right ? |
19:18.23 | terrapen | yup, and PoE |
19:18.32 | brad_mssw | eh, don't care about PoE |
19:18.41 | terrapen | we just ordered a 48-port Foundry 10/100/1000 PoE switch today |
19:18.46 | *** join/#asterisk brockj49464_ (n=brockj49@41.105.dhcp.hope.edu) |
19:19.01 | [TK]D-Fender | Well I hate the fact that linksys.com hides the entire line from public view... says something about them... |
19:19.08 | [TK]D-Fender | terrapen : $? |
19:19.11 | brad_mssw | terrapen: we use all HP Procurve switches here |
19:19.18 | malverian | I forgot to ask my question ;) |
19:19.21 | brad_mssw | terrapen: btw, where did you buy your 942s ? |
19:19.22 | [TK]D-Fender | terrapen : must be NASTY to have 1000 as well |
19:19.27 | terrapen | voipsupply |
19:19.38 | timscott | Do any of you have suggestions regarding SIP and NAT? I'm trying to associate a phone inside of a NAT with a server inside of a different NAT. |
19:20.10 | [TK]D-Fender | timscott : WIKI tells you what to do. |
19:20.19 | timscott | voip-info.org, you mean? |
19:20.26 | malverian | If I'm using call parking, what's the best way to make sure CDR is handled sanely? I have an extension (say 600) that is a call parking extension.. when a representative transfers a customer there, and then a different representative picks up the person from that parking spot, I want to see two separate CDR entries. |
19:20.32 | malverian | Any recommendations on how to accomplish this? |
19:20.39 | terrapen | I'm geting 10/100/1000 because I'm going to re-purpose this switch when our big Foundry chassises arrive this summer |
19:21.04 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
19:21.12 | Leob | can anyone help me with ODBC? |
19:21.22 | malverian | Currently, when I do this, all of the CDR variables are lost when the customer is transfered, so I lose the "dst", "src" etc variables. |
19:21.24 | terrapen | tk, the 942's list at $199 but are cheaper in bulk |
19:22.33 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
19:23.05 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
19:23.24 | malverian | So I guess my question is.. how can I make CDR variables carry over across blind transfers? |
19:23.54 | [TK]D-Fender | terrapen : Better off getting sperate switches and picking your jacks... because anywhere you will use PoE can't support GBIT at the same time... better off cross-conencting. Keeps your cost MCUH lower |
19:24.35 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
19:27.35 | [TK]D-Fender | terrapen : lets say 25 phones.. how low could you go (in retail, no "extra" special pricing) |
19:28.25 | [av]bani | [TK]D-Fender: gbit works with poe |
19:29.34 | [TK]D-Fender | [av]bani : Never heard of it at the same time... PoE uses 2 pairs w/o data |
19:30.01 | [TK]D-Fender | [av]bani : and every device quoting it seems to say that GBIT only applies if its wall-warted |
19:30.03 | [av]bani | [TK]D-Fender: there's literally billions of gbe-poe products... |
19:30.22 | [TK]D-Fender | [av]bani : Got a decent link to one that well douments this ability? |
19:30.26 | [av]bani | http://www.google.com/search?q=%2B%22gigabit%22+%2B%22poe%22&hl=en&lr=&start=10&sa=N |
19:31.00 | Shaun2222 | what does this do? ignorepat => 9 |
19:31.11 | timscott | you press 9, it doesn't break the dialtone. |
19:31.17 | Shaun2222 | ok |
19:31.35 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
19:31.36 | malverian | I wish there was a better explanation of how CDR works.. |
19:32.00 | Hmmhesays | freaking IIS |
19:32.21 | [TK]D-Fender | [av]bani : Can you link me a specific product... I hate the hunting... |
19:32.26 | Shaun2222 | timscott: doesnt the phone send the dialtone? mine does i think, i dont see a connection to asterisk until it times out and attempts to dial or i press dial |
19:32.42 | timscott | it depends how you're using it, yeah. |
19:32.53 | Shaun2222 | can that be changed? |
19:32.57 | timscott | if your phone is generating the dialtone, then it doesn't really matter if you hit "9" or not |
19:33.08 | timscott | because your phone is waiting for the whole string to be completed before sending the string |
19:33.22 | timscott | yeah, I guess, depending on your phone, there might be a setting for early dialing |
19:33.28 | timscott | or something similar |
19:33.39 | Shaun2222 | i'm using the cisco 7960 phone, is their a better phone to use with asterisk... this phone seams kind of lame and it seams almost like asterisk and this phone lack the features of a real pbx. |
19:33.46 | [av]bani | [TK]D-Fender: http://www.eeproductcenter.com/powersources/brief/showArticle.jhtml?articleID=168602163 |
19:33.54 | timscott | I'm not sure, you'd have to ask someone else about that phone |
19:34.01 | timscott | I'm not familier with cisco products, really. |
19:34.03 | timscott | sorry mate. |
19:34.04 | [av]bani | [TK]D-Fender: http://cs.pennnet.com/Articles/Article_Display.cfm?Section=ONART&cat=INDUS&p=42&ARTICLE_ID=225388&VERSION_NUM=1 |
19:34.17 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
19:34.19 | Shaun2222 | timscott: what type of phones do you use? |
19:34.31 | h3x0r | sha |
19:34.38 | h3x0r | sha: the 79xx series was designed to use with ccm |
19:34.45 | *** part/#asterisk Grizzy (i=Generic@ppp-71-133-231-94.dsl.pltn13.pacbell.net) |
19:35.03 | Shaun2222 | h3x0r: yes, is that the reason, is their a better phone? |
19:35.05 | [av]bani | [TK]D-Fender: http://www.powerdsine.com/news/prs/pr_030505_Gigabit_Midspan.html |
19:35.14 | jsharp | If you're waiting for timeout to dial on your 7960s, then you don't have a proper dialplan set for it. |
19:35.17 | timscott | Shaun: Grandstream GXP2000, and Snom190, and an SPA3000 |
19:35.36 | timscott | jsharp: I think it was just an example. |
19:35.40 | h3x0r | snom 320 and 360 is the way to go |
19:35.49 | h3x0r | 190 is a elmeg phone |
19:35.50 | Shaun2222 | jsharp: i could change it in the dialplan sure, but so far i dont see the point in even having a phone with a tone of buttons... |
19:35.52 | timscott | If you can afford it, h3x0r. |
19:35.54 | [av]bani | snom 360 .. lol |
19:35.58 | timscott | We aren't all made of money, like you. |
19:36.00 | h3x0r | they dont cost much more than the other phones |
19:36.02 | h3x0r | the 320 is nice |
19:36.07 | timscott | Mmhmm, only like 100 bucks more. |
19:36.18 | timscott | whatever |
19:36.18 | [av]bani | h3x0r: the snom hardware is ok, the firmware is *horrible* |
19:36.20 | h3x0r | no like 40 bucks more |
19:36.26 | timscott | yes, right, you win. |
19:36.27 | [av]bani | i would not recommend snom 3xx toanyone |
19:36.28 | h3x0r | the new firmware is better |
19:36.32 | [av]bani | no, it isn't. |
19:36.35 | h3x0r | they fixed most of the bugs |
19:36.37 | [av]bani | nope. |
19:36.43 | h3x0r | whens the last time you upgraded |
19:36.43 | timscott | 190's firmware seems alright. haven't had any real big problems with it |
19:36.44 | *** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-94.dsl.pltn13.pacbell.net) |
19:36.45 | [av]bani | it still locks up. still has horrile UI |
19:36.51 | [av]bani | hmm... last week? |
19:37.05 | [av]bani | running 5.5.1b |
19:37.10 | *** join/#asterisk boch (n=fran@unirc.com.ar) |
19:37.16 | boch | hi guys, need your help to set the pppoe password via the ivr menu of a sipura spa-2100 |
19:37.27 | [av]bani | it still locks up, still has a HORRIBLE ui, still has incorrect US indications |
19:37.33 | jsharp | I like the 7940s we have. Once we got over the heartburn of reflashing them to SIP, that is. |
19:37.33 | [av]bani | still has 90% of the bugs I reported to snom MONTHS ago |
19:38.25 | h3x0r | like what |
19:38.37 | [av]bani | why? can you fix them? |
19:38.41 | *** part/#asterisk justinu|laptop (n=Justin@66.209.15.236) |
19:39.03 | timscott | bani: what's wrong with the 3xxx's that isn't wrong with the 190's? |
19:39.06 | timscott | err |
19:39.09 | timscott | *3xx's |
19:39.12 | [av]bani | timscott: dunno, i dont have a 190 |
19:39.25 | timscott | :/ |
19:39.25 | [av]bani | but the 3xx is poo |
19:39.28 | h3x0r | they dont sell 190s anymore |
19:39.40 | timscott | my only real bug with the 190 is that it's kind of light...it slides around on my desk. :) |
19:39.44 | timscott | Other than that, they're fine |
19:39.56 | Shaun2222 | what does include => default load? |
19:40.08 | timscott | the [default] context, if there is one |
19:40.15 | h3x0r | snom fixed the bugs i bitched about fairly promptly |
19:40.26 | [av]bani | h3x0r: snom just blew off my last bugreport, saying it's a deliberate feature and won't ever be fixed. |
19:40.46 | Shaun2222 | i see |
19:40.46 | h3x0r | and whats that |
19:40.52 | timscott | bani: what's that? |
19:41.11 | [av]bani | on the idle screen, if you adjust ringer volume, it plays ringer 4 always. regardless of the actual ringer you have configured (eg custom ringtone, with completely different volume than the phone's internal ringer 4) |
19:41.12 | timscott | Shaun: By the way, don't take _anything_ that I say as 100% truth and fact... |
19:41.22 | timscott | Honestly, I don't know half the time, it's just educated guessing. |
19:41.26 | [av]bani | so the ringtone voluem on the idle screen will likely not be ANYWHERE close to the actual ringer. |
19:41.37 | h3x0r | and this is the latest 5.x firmware? |
19:41.44 | [av]bani | yes. and snom says they won't fix it. |
19:41.48 | timscott | Oh snap! |
19:41.50 | [av]bani | they say it's a deliberate decision. |
19:42.07 | h3x0r | how does that interfere with you using your phone |
19:42.11 | timscott | bani, you're obviously a purist. ;) |
19:42.29 | [av]bani | h3x0r: because the ring voluem adjustment on the idle screen has NOTHING to do with the actual ring volume. |
19:42.38 | [av]bani | it is INCORRECT, not only playing the wrong ringtone, but the wrong volume |
19:42.50 | [av]bani | now if you want things that interfere with me using my phone... |
19:42.53 | h3x0r | the only thing that pisses me off on the 360 is the font size |
19:42.55 | [TK]D-Fender | timscott : No, he just has the worlds largest raging bile ducts EVER. |
19:42.56 | [av]bani | it LOCKS UP all the time |
19:43.03 | h3x0r | and they told me its that way to display unicode characters |
19:43.06 | [av]bani | yeah, the font size is annoying too. |
19:43.14 | h3x0r | so thats why i use 320s :P |
19:43.20 | [av]bani | 90% of my caller IDs are cutoff |
19:43.21 | timscott | D-Fender: :D |
19:43.23 | h3x0r | the larger screen on the 360 is useless |
19:43.27 | [av]bani | hell, snom's OWN MENUS get cut off... |
19:43.41 | timscott | Wow, I'm so glad I didn't buy a 3xx. |
19:43.43 | timscott | 190 <3 |
19:43.54 | [av]bani | you can't force the backlight on either. |
19:43.55 | h3x0r | the 320 has 14 line keys |
19:43.58 | h3x0r | enough said |
19:43.58 | timscott | Next time I get the chance, I'll try one out and see for myself. |
19:44.01 | [av]bani | snom also said they won't fix that. |
19:44.15 | h3x0r | i wouldnt leave a electronumicandescent backlight on all the time either |
19:44.21 | [av]bani | the US indications are wrong. |
19:44.21 | h3x0r | it would burn out in a couple years |
19:44.24 | austinnichols102 | snom is a stupid name |
19:44.29 | timscott | haha zing! |
19:44.43 | h3x0r | snom 3xx sucks but everything else sucks more |
19:44.45 | [av]bani | but the biggest annoyance is the phone locks up all the time... |
19:44.53 | [av]bani | no, i have a cisco and it is very nice. |
19:45.02 | timscott | I've never had a problem with locking up on any of my phones... |
19:45.03 | [av]bani | hell, even the grandstreams don't f*cking lock up |
19:45.08 | [av]bani | only snom does |
19:45.09 | h3x0r | mine has never locked iup |
19:45.10 | h3x0r | up |
19:45.13 | h3x0r | maybe yours is defective |
19:45.17 | [av]bani | nope |
19:45.23 | [av]bani | and... lots of other people have it lock up too |
19:45.29 | [av]bani | it locks up mostly on transfers or conf calls |
19:45.30 | timscott | Flame war! |
19:45.37 | timscott | Snom sucks! |
19:45.39 | timscott | No it doesnt! |
19:45.41 | timscott | Yes it does! |
19:45.43 | timscott | No it doesnt! |
19:45.45 | timscott | whatever guys,. |
19:45.47 | [av]bani | it's not an isolated problem with defective hardware. snom knows about it, they've acknowledged the bug |
19:45.52 | [av]bani | they keep saying it's fixed, but it's not |
19:45.54 | h3x0r | at least it does conference calls right |
19:46.17 | [av]bani | or rather, they keep promising it's fixed in the latest firmware. but we all keep having lockups anyway |
19:46.33 | h3x0r | you are using 5.3.6? |
19:46.59 | [av]bani | i've tried everything from 5.0 to 5.5.1b |
19:47.20 | h3x0r | 5.5 ?! |
19:47.27 | [av]bani | yes |
19:47.32 | msw | anyone know what might give me "Cause: Invalid number format (28)" on my PRI? |
19:47.37 | msw | (this is my initial bringup) |
19:47.44 | h3x0r | where do you get that |
19:47.51 | msw | I've tried national and unknown pridialplan |
19:47.59 | [av]bani | snom really broke 5.4 also... totally broke autoprovisioning |
19:48.02 | msw | h3x0r: pri debug span N |
19:48.09 | [av]bani | then blamed me for having "incorrect programming" |
19:48.11 | brif8 | Is there a way to route calls from one * box to another via the Internet where both * boxes are behind NAT ? |
19:48.13 | h3x0r | no i mean snom firmware 5.5 |
19:48.16 | timscott | brif: yes |
19:48.24 | [av]bani | i sent them screenshots which proved it was their fau |
19:48.26 | timscott | That's the problem I am actually working on at this very moment... >_< |
19:48.26 | [av]bani | fault |
19:48.32 | msw | h3x0r: sorry, that wasn't for me, was it? |
19:48.36 | timscott | brif: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
19:48.37 | [av]bani | getting them to acknowledge the bug and fix it was a major PITA |
19:48.39 | h3x0r | nope msw |
19:48.49 | timscott | brif: http://www.voip-info.org/wiki/view/NAT+and+VOIP |
19:49.09 | sevard | So I have a question about register times |
19:49.23 | Katty | the answer is 42 |
19:49.26 | h3x0r | are you using taht phone on the public internet by chance |
19:49.33 | sevard | If * gets rebooted (and it does need to rebooted often) the phone stops ringing although * calls it |
19:49.39 | sevard | Katty: six by nine? |
19:49.39 | timscott | brif: hope that helps, it's where I'm at right now... :/ |
19:49.53 | Katty | sevard: this is not algebra. |
19:49.53 | timscott | No, 6 by 7. |
19:50.03 | Katty | sevard: besides, that's 54 |
19:50.07 | [av]bani | h3x0r: it's obvious that snom has never tested the 360 outside the confines of their german offices. |
19:50.20 | sevard | Katty: I was just at a trade show demoing voip, i named one trillian, one arthur, one marvin, and one peter jones, phones. and nody got my theme. |
19:50.28 | sevard | obviously you guys don't get the six by nine either :(* |
19:50.28 | timscott | hahaha |
19:50.34 | timscott | No, sorry. |
19:50.39 | timscott | I only got the funny joke. |
19:50.40 | timscott | :S |
19:50.41 | sevard | oh, another phone was Ford |
19:50.45 | timscott | What's the six by nine all about? |
19:50.58 | timscott | If you don't tell me, I'm going to have to waste my time googling. :S |
19:50.58 | sevard | that's the ultimate question |
19:51.03 | sevard | what is six by nine |
19:51.14 | sevard | the ultimate answer is 42 |
19:51.14 | timscott | oh right |
19:51.16 | timscott | durr |
19:51.17 | timscott | sorry |
19:51.22 | [av]bani | h3x0r: http://www.snom.com/wiki/index.php/Beta_Firmware |
19:51.27 | sevard | bbiab have to move my car so the parking bitch doesn't slaughter me |
19:51.31 | h3x0r | yea i just found it on the wiki |
19:51.31 | msw | nevermind - bad Dial() cmd |
19:51.37 | timscott | forgot, base 13 |
19:51.57 | jsharp | msw: Sending alpha characters onto the PRI? |
19:52.30 | h3x0r | http://www.voip-info.org/wiki/view/snom+360 |
19:52.32 | h3x0r | haha |
19:52.34 | *** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-031.mycingular.net) |
19:52.34 | h3x0r | its the bug list |
19:52.39 | timscott | Guys. Who cares. |
19:52.51 | Katty | i'm feelin pretty lost right now |
19:52.56 | Katty | but that's ok! |
19:52.59 | Katty | i'm used to it. |
19:53.00 | timscott | youre in #asterisk on freenode.ent |
19:53.02 | timscott | *net |
19:53.03 | Hmmhesays | i'm feeling like i want to knock some head |
19:53.06 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
19:53.20 | Katty | oh thank goodness. |
19:53.22 | [TK]D-Fender | Katty : There, you're found! |
19:53.22 | Katty | i'm no longer lost. |
19:53.24 | Katty | k, all better. |
19:53.27 | timscott | *follows |
19:53.30 | timscott | :( |
19:53.36 | Katty | Hmmhesays: i'm up for a little sparing |
19:53.53 | [av]bani | h3x0r: that's not all the bugs, just the ones i was annoyed enough to post publically because snom ignored me |
19:54.00 | msw | jsharp: it wasn't stripping off the prefix properly |
19:54.10 | techman97_andy | hey all - question. My SIP provider allows me to change the codec from uLAW to G729A. I did that, but all of my calls were inaudible...scratching, popping, and the CLI was going nuts with errors. I thought I saw G729A in the list of supported codecs that came with Asterisk - any ideas? |
19:54.45 | [av]bani | h3x0r: mike240se is one of the other people having lockups with his snom phones... if it is "defective hardware" then snom really has severe QC issues with their manufacturing. |
19:54.59 | [TK]D-Fender | Hmmhesays : Yeah... spare her ;) |
19:55.09 | h3x0r | its like a german car huh :) |
19:55.16 | RaYmAn-Bx | techman97_andy: it's only supported in pass-through mode unless you buy a G729 license |
19:55.32 | *** join/#asterisk badboyz (n=will@216.87.37.130.primary.net) |
19:55.46 | [av]bani | h3x0r: also.. the snom 360 runs REALLY warm... all other phones I have don't get anywhere near that hot |
19:55.53 | [av]bani | even my cisco 7970 |
19:55.58 | techman97_andy | so if I had SIP phones hanging off of Asterisk going to this SIP provider in G729A, I would need a license for every channel? |
19:56.04 | badboyz | so if i have a t1, and i want to connect it to an asterisk box, to use as a backup for outgoing phone calls... which interface card do i need? |
19:56.21 | sevard | Alright, so |
19:56.33 | [TK]D-Fender | techman97_andy : Every channel in use at a given time. But thats only in places where * neds to inject audio (like in the voicemail system, or IVR's, etc) |
19:56.33 | sevard | If * gets rebooted (and it does need to rebooted often) the phone stops ringing although * calls it |
19:56.55 | sevard | It sounds like a dumb idea for the phjone to re register to the server every 60 seconds to fix this problem |
19:57.00 | sevard | phone* |
19:57.10 | sevard | but.. that's the only answer I can come up with. |
19:57.33 | techman97_andy | ok - so my SIP provider only has one channel that I go out...although it has multiple pipes that are accessible...9 SIP phones, and 1 IVR...would that by two G729A licenses? |
19:57.43 | h3x0r | anybody use that linksys cordless voip phone yet |
19:57.57 | [av]bani | h3x0r: the snom 360 is a real letdown for such an expensive phone. other phones in the same price bracket are 100x better. |
19:58.11 | [av]bani | and most of the issue is with snom's shitty firmware. |
19:58.12 | h3x0r | like what |
19:58.14 | h3x0r | ciscos phones suck |
19:58.21 | timscott | uhh |
19:58.23 | timscott | heathen? |
19:58.24 | h3x0r | polycoms you cant legitamtely use with asterisk |
19:58.27 | [av]bani | i have cisco, grandstream, polycom, and snom |
19:58.34 | [av]bani | um what? |
19:58.38 | mog_work | why cant you use polycoms? |
19:58.38 | h3x0r | grandstream is awful |
19:58.40 | *** join/#asterisk exonic (n=exonic@sig.triton.net) |
19:58.44 | sevard | h3x0r: I just got an Aastra 408i CT and it's everything I'ev ever wanted in any phgone ever. |
19:58.46 | [av]bani | polycom has no problems using with asterisk. |
19:58.47 | sevard | phone* |
19:58.51 | timscott | Aastra = awesome. |
19:58.53 | h3x0r | its not "legal" |
19:58.54 | [av]bani | in fact asterisk is an _officially supported platform_ by polycom |
19:58.55 | timscott | Aastra for the win. |
19:59.04 | tainted- | why can't u use polycom |
19:59.04 | [av]bani | h3x0r: um. sorry, but... BS |
19:59.04 | mog_work | what do you mean its not legal? |
19:59.06 | mog_work | polycom supports astersik |
19:59.08 | mog_work | err asterisk |
19:59.09 | exonic | Hey folks, What' s agood solution to peer asterisk with 4+ lines? I would like to integrate into FXO ports that are already in existance. |
19:59.09 | [av]bani | it's perfectly legal, and even supported. |
19:59.10 | mog_work | as guess what |
19:59.14 | sevard | timscott: This phone is _so god damn awesome_ |
19:59.18 | mog_work | people want to buy phones to use with asterisk. |
19:59.20 | timscott | sevard: that's what I've heard. |
19:59.22 | timscott | I want to get one. :D |
19:59.25 | mog_work | more than they want a polycom pbx and phones |
19:59.30 | h3x0r | <PROTECTED> |
19:59.33 | h3x0r | voipsupply |
19:59.34 | h3x0r | atacomm |
19:59.35 | h3x0r | etc. |
19:59.36 | [av]bani | h3x0r: i have no idea where you heard that, but it's bullshit |
19:59.37 | Netgeeks | I like my cisco phone, it has worked perfectly for me for years and I spend many hours a day on the phone... |
19:59.37 | tainted- | sevard what phone |
19:59.37 | mog_work | you are wrong |
19:59.42 | mog_work | its total bullshit |
19:59.44 | sevard | timscott: more expensive than most but BY FAR _the best_ phone I've ever used ever. |
19:59.52 | sevard | tainted-: Aastra 480i CT |
19:59.54 | mog_work | polycom sent several phones to digium |
19:59.57 | mog_work | to be sure they all worked |
20:00.00 | tainted- | sevard i heard it's shit |
20:00.01 | mog_work | i have one on my desk |
20:00.04 | justinu|laptop | i'v never had that problem with anyone |
20:00.04 | [av]bani | h3x0r: um. voipsupply and atacomm know perfectly well polycom is legal to use with asterisk. |
20:00.07 | mog_work | kevin is talking on one right now |
20:00.07 | h3x0r | ok |
20:00.08 | timscott | Aastra built stuff for TDM pbxs before they built IP, mmhmm. |
20:00.12 | timscott | unless I am mistaken |
20:00.13 | [av]bani | h3x0r: voipsupply even advertises polycoms for that purpose. |
20:00.13 | sevard | tainted-: My review is it's the absolute best. |
20:00.15 | h3x0r | maybe they just dont make money on them |
20:00.22 | mog_work | heh |
20:00.23 | mog_work | sure.......... |
20:00.24 | sevard | timscott: you're not mistaken |
20:00.31 | timscott | they've had a while in the industry to learn how things should work |
20:00.35 | tainted- | sevard have u used snoms, polycoms, or ciscos? maybe u have nothing to reference to |
20:00.37 | sevard | It doesn't feel cheap, it feels like a real phone, it has NO draw backs o that I can tell |
20:00.46 | timscott | 'cept $$$++ |
20:00.48 | timscott | :D |
20:00.50 | h3x0r | the cisco 94x phones are awful |
20:00.51 | timscott | But it's worth it |
20:00.55 | tainted- | sevard that's the one with the xmlbrowser right? |
20:00.56 | h3x0r | and the legal issue with 79xx applies |
20:01.01 | sevard | I've used snom and polycoms and used a HOP1--2 for a long time |
20:01.03 | sevard | HOP 1002 |
20:01.09 | austinnichols102 | what's the 79xx legal issue? |
20:01.10 | h3x0r | or i should say, the fact that they suck with sip |
20:01.17 | Netgeeks | legal issues with the 79xx? |
20:01.25 | sevard | tainted-: I haven't dug into xml on this phone yet |
20:01.25 | h3x0r | you have to pay license fees to use the phone |
20:01.25 | jsharp | SIP licensing and all that. |
20:01.38 | [av]bani | there's no licensing issues. i got that direct from cisco. |
20:01.51 | austinnichols102 | no problem with sip licensing. |
20:01.52 | [av]bani | the only license is for ccm |
20:01.52 | h3x0r | regardless the sip firmware for 79xx is a joke |
20:01.57 | Netgeeks | you buy the phone for a authorized cisco reseller, and you are fine |
20:02.08 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
20:02.08 | austinnichols102 | I'm not laughing |
20:02.18 | Netgeeks | um okay h3x0r |
20:02.29 | h3x0r | pots phones on an ata do way more than a 7960 |
20:02.43 | brodiem | anyone have experience with using fax w/ ATA (specifically SPA1001)? |
20:02.46 | austinnichols102 | example? |
20:02.54 | malverian | If I do a "NoCDR" sometime earlier on in the dialplan.. will ResetCDR undo that? |
20:02.57 | [av]bani | h3x0r: no, you don't. |
20:02.59 | h3x0r | not being able to drop a conference party |
20:03.11 | [av]bani | the only license you need is for CCM. please stop. |
20:03.35 | h3x0r | you need cco to download the sip firmware |
20:03.48 | h3x0r | to use cco means you are supposed to buy the support contract on your phone |
20:04.06 | austinnichols102 | they charge for fw updates, so what... |
20:04.44 | h3x0r | updates? they ship with SCCP firmware |
20:05.39 | [av]bani | h3x0r: there are various support contracts, the cheapest one is $8 |
20:05.52 | Netgeeks | so is your dislike for the 79xx phones based upon cisco business practices, or do you actually not like the way the phone functions, or both? |
20:06.30 | timscott | Is anyone here at all familier with SIP and NAT workarounds for Snom190's? |
20:06.32 | austinnichols102 | I would like to have keepalive and a couple of other features in the phone, but I'm far from not liking the phone |
20:06.47 | timscott | Is anyone here at all familier with SIP and NAT workarounds for Snom190's? |
20:06.48 | timscott | >_< |
20:07.28 | [av]bani | cisco is pretty lax with firmware policies though. you can usually call them up without any support contract and ask for the latest sccp and they'll send it along. |
20:07.35 | brif8 | can I still join two * boxes with SIP if the extensions are the same on both? I basically want to route all incoming calls on Box 1 to [incoming] on box 2 |
20:07.40 | [av]bani | (they will also do this with ios, when there are security alerts) |
20:08.27 | [av]bani | so um h3x0r, what phones do you actually have besides a snom 3xx? |
20:09.12 | timscott | sooo...I'm guessing no one, then. :'( |
20:09.16 | [TK]D-Fender | brif8 : You can do most things you can imagine with * |
20:09.17 | sevard | I strongly suggest getting an aastra 480i ct from a retailer that you can return it to if you don't like it |
20:09.21 | brif8 | I'm reading WIKI but with the day I've had it isn't making much sense http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
20:09.42 | [TK]D-Fender | brif8 : you can send calls from anywhere to anywhere.. its up to you. |
20:10.23 | brif8 | [TK]D-Fender: care to please just clarify what I need to do. I tried simply adding a extension on box 1 to DIAL(SIP/nex@serverBIP,80,Ttr) but it rings once and fails |
20:10.32 | malverian | I need some help with a CDR problem I'm running into.. |
20:10.39 | brif8 | I probably need to have a register => but I'm not sure |
20:10.45 | malverian | If anyone has a moment to give me some pointers, I'd greatly appreciate it. |
20:10.54 | *** join/#asterisk ToTo (n=ToTo@host55-145.pool870.interbusiness.it) |
20:11.50 | Dandan | IAX2 to SIP/broadvoice translation doesn't work... |
20:11.50 | Dandan | no errors reported |
20:11.52 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
20:11.53 | timscott | I would also greatly appreciate it if someone would take a moment to give me pointers :S |
20:11.57 | malverian | The scenario is like so: 1) Customer calls in to main line and a sales rep answers the phone, 2) Sales rep puts the customer in "parking lot", 3) Another sales rep picks up the customer from parking lot |
20:13.14 | malverian | The following call detail records get logged: 1) src = customer, dst = rep, 2) src = rep, dst = parking lot |
20:13.53 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
20:14.06 | malverian | I need #2 to instead show src = customer, dst = rep. The Set(CDR(src)=....) functions say that the CDR variable is read only.. do I have any chance here? |
20:14.18 | Katty | what's better, the ip500 or the 501? |
20:14.30 | Katty | are there any major differences? |
20:14.35 | Katty | did the 501 sprout wings? |
20:14.38 | Katty | does it do a little dance? |
20:14.47 | timscott | hmm |
20:14.57 | timscott | sometimes the "X01"s have a second port |
20:14.59 | timscott | a mini-hub |
20:15.02 | malverian | timscott, I have no idea.. there is a wiki on the SNOM site, have you checked tha tout? |
20:15.15 | timscott | malverian: reading it at the moment :S |
20:15.18 | brodiem | what is the most reliable way of using a physical fax machine? Getting about 90% fail rate using SPA1001.. |
20:15.25 | timscott | haha, use POTS |
20:15.33 | timscott | brodiem :S |
20:15.39 | malverian | brodiem, Install FXS card, run analog cable :-P |
20:16.04 | [TK]D-Fender | brif8 : That Dial sample sends an UNAUTHENTICATED call to the other server, which would have to be set up to accept them. usually an unsafe thing when "forwarding" calls... |
20:16.13 | brodiem | malverian, my only worry is that Digium doesn't recommend their FXS for faxing because of the timing device |
20:16.31 | malverian | brodiem, Does your trunk have a timing device? |
20:16.39 | brif8 | [TK]D-Fender: I accept that but this is just a temporary test |
20:16.46 | [TK]D-Fender | Katty, timscott : Incorrect. X01's have more ram to support larger SIP & BR images |
20:16.52 | brif8 | how would I authenticate two * boxes |
20:16.54 | malverian | brodiem, We have a PRI here, and our fax machines work almost 100% successful with wildcards. |
20:16.59 | Katty | [TK]D-Fender: thanky. |
20:17.04 | timscott | Fender ok |
20:17.09 | [TK]D-Fender | brif8 : then you need to have a context set in [general] and allowguest=yes in sip.conf |
20:17.16 | gandhijee | Dandan: what do you mean it doesn't work? |
20:17.25 | [TK]D-Fender | Katty : and in the case of the 601, the ability to add on the attendant modules. |
20:17.27 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
20:17.31 | timscott | I just ate like a whole salmon |
20:17.39 | brodiem | malverian, yes I believe so (at least I have the telco setup as the master timing source in zaptel.conf) |
20:17.44 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
20:17.46 | gandhijee | Dandan: are you sure you are registered w/ broadvoice? |
20:17.47 | Netgeeks | Teh IP501 sprouts wings? |
20:17.51 | Netgeeks | The, even? |
20:17.53 | malverian | brodiem, Then you should be fine. |
20:18.01 | [TK]D-Fender | timscott : I doubt that unless you got a tiny baby one... which typically would get thrown back... |
20:18.02 | malverian | brodiem, We had someone come test the line and we had no timeslips. |
20:18.10 | brodiem | malverian, cool |
20:18.11 | timscott | dude it was like three lbs of salmon |
20:18.15 | timscott | i'm stuffed |
20:18.19 | timscott | with salmon. |
20:18.21 | brodiem | malverian, is there a single FXS card or just the 400p w/ module available? |
20:18.23 | Dandan | gandhijee: yeah |
20:18.23 | Dandan | i can do sip-sip through bv |
20:18.24 | Dandan | no probs |
20:18.25 | malverian | brodiem, Er.. frame slips. |
20:18.34 | [TK]D-Fender | timscott : Do you have any real idea just how big a mature salmon gets? |
20:18.38 | mog_work | malverian, any chance you might finish you sphynx pre 1.4? |
20:18.41 | malverian | brodiem, I'm not certain.. I _think_ there are dual span. |
20:18.41 | gandhijee | Dandan: what version you running? |
20:18.53 | malverian | mog_work, Yes, as soon as I figure out how to fix this damn CDR issue :-P |
20:18.55 | timscott | Fender: Relax, man...it's just talking. |
20:19.03 | gandhijee | Dandan: i was using my asterisk box w/ broadvoice till last week, has IAX, MGCP and SIPs phones |
20:19.06 | gandhijee | no problems |
20:19.10 | [TK]D-Fender | brodiem : TDM for FXS is pretty wasteful $ wise, and less functional / portable |
20:19.27 | malverian | [TK]D-Fender, But more reliable for faxing in my experience here. |
20:19.31 | Leob | NOVICE QUESTION: can anyone help me configure ODBC? I've been trying for about 8 hours and I have no idea of how to check if I'm doing something wrong... Any suggestion would be more than welcome!! |
20:19.34 | mog_work | whats wrong with cdr |
20:19.42 | [TK]D-Fender | malverian : than a decent ATA? |
20:19.43 | brodiem | [TK]D-Fender, what is the alternative for fax? besides a completely seperate pots |
20:19.46 | malverian | mog_work, Just a problem with a specific thing I am trying to do with it. |
20:19.50 | Dandan | 1.2.6 |
20:19.50 | Dandan | i use cubix |
20:19.50 | Dandan | as iax |
20:19.51 | Dandan | to call sip via bv |
20:19.51 | Dandan | if i do iax - sip internally all is good |
20:19.52 | malverian | [TK]D-Fender, Ah.. no. |
20:19.55 | malverian | [TK]D-Fender, But that's usually overkill. |
20:19.57 | [TK]D-Fender | malverian : OH, faxing... nevermind ;) |
20:20.02 | brodiem | lol |
20:20.03 | [av]bani | hmm.. gxp2000 does rfc2833 fine, polycom 601 doesn't... |
20:20.04 | [av]bani | weird |
20:20.08 | austinnichols102 | anyone familiar with US LEC as a carrier? I was shocked to find out that they support asterisk. |
20:20.13 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
20:20.17 | malverian | austinnichols102, Yeah, we use them in Gainesville. |
20:20.18 | [TK]D-Fender | [av]bani : Mine seems to be fine here with RFC... |
20:20.18 | Dandan | austinnichols102: I am |
20:20.27 | Dandan | actually my current provider is taking them over |
20:20.30 | malverian | austinnichols102, They are the worst customer service I've ever epxerienced :-P |
20:20.34 | malverian | *experienced |
20:20.37 | gandhijee | i used idefisk as my IAX client, a Yxtn phone w/ MGCP and a snom 360 w/ sip, no problems |
20:20.42 | [av]bani | [TK]D-Fender: i can't get our 601's to do dtmf properly. gxp2k's work fine. so it's not an asterisk config problem. |
20:20.44 | malverian | austinnichols102, Their "professionals" are complete noobs. |
20:20.48 | austinnichols102 | great |
20:20.49 | timscott | happy noob |
20:20.50 | [av]bani | [TK]D-Fender: and i'm using the stock sip.cfg from polycom... |
20:21.05 | brodiem | malverian so a $140 tdm10b is probably the cheapest way to get analog fax? |
20:21.15 | gandhijee | Dandan: taking who over? |
20:21.15 | h3x0r | lets see i have cisco 7960, grandstream gxp2000, snom 320 and 360, linksys 941, sipura 841, |
20:21.23 | Dandan | us lec |
20:21.32 | gandhijee | is taking over broadvoice? |
20:21.34 | Dandan | ctc is taking us lec over |
20:21.35 | Dandan | noooo |
20:21.41 | [TK]D-Fender | [av]bani : I use basic stock setups too.. never a problem, and I use them at home a lot now too. |
20:21.58 | [av]bani | [TK]D-Fender: well i'm stuck, i can't figure out why the 601 isn't sending dtmf. sucks :( |
20:22.07 | [TK]D-Fender | [av]bani : I hear ya... |
20:22.13 | [av]bani | [TK]D-Fender: i keep bashing my head against stupid polycom documentation |
20:22.20 | gandhijee | speakin of the 601's is there an XML editor for those bitches? |
20:22.25 | austinnichols102 | who's CTC? |
20:22.27 | [av]bani | no |
20:22.28 | gandhijee | or an easier way to edit that file |
20:22.34 | [av]bani | nope |
20:22.38 | gandhijee | =( |
20:22.38 | h3x0r | i sold the other stuff i had |
20:22.40 | [TK]D-Fender | gandhijee : there are a number of XML editors out there.... |
20:22.43 | h3x0r | i should try polycom |
20:22.59 | [TK]D-Fender | h3x0r : I did the same... to BUY my Polycom's :) |
20:23.01 | [av]bani | polycom are harder to provision than even cisco. |
20:23.13 | malverian | mog_work, I queried you to avoid sending repeat information to the channel. |
20:23.13 | gandhijee | yeah they are |
20:23.20 | [av]bani | you keep finding these stupid little hidden undocumented switches in the configs |
20:23.21 | timscott | I thought ciscos were nice to provision |
20:23.32 | timscott | Can't you import a CSV file to configure them? |
20:23.34 | [av]bani | timscott: not with asterisk |
20:23.35 | gandhijee | but they sound great |
20:23.38 | [TK]D-Fender | [av]bani : Maybe a little slower on the start, but once you know, I doub't cisco is really much better... |
20:23.48 | [av]bani | [TK]D-Fender: cisco is a lot simpler |
20:23.56 | [av]bani | theres no 5 billion config entries |
20:24.01 | timscott | bani: Can't you import a CSV file into most cisco phones for provisioning? |
20:24.05 | [av]bani | polycom is a real mess |
20:24.06 | timscott | Makes for easier automation |
20:24.12 | [TK]D-Fender | [av]bani : I dunno... I got my news ones done from scratch in no time flat... |
20:24.14 | gandhijee | you'd think polycom would put out a file to help you config them easier, specailly since they "Now Support Asterisk" |
20:24.17 | Dandan | Ctc communications |
20:24.23 | Dandan | www.ctcnet.com |
20:24.26 | [av]bani | [TK]D-Fender: because your config is simple and you had a ready made template. |
20:24.27 | austinnichols102 | tks |
20:24.39 | [av]bani | [TK]D-Fender: someone coming to polycom from scratch isn't nice |
20:24.40 | Dandan | they are getting bigger by the month... |
20:24.53 | [av]bani | out of the box they are a major pita |
20:24.59 | austinnichols102 | dandan: considering us lec, but don't have time to really deal with more carrier-level stupidity |
20:25.33 | gandhijee | i still don't understand everything bout those phones |
20:25.36 | h3x0r | you know whats crazy |
20:25.37 | austinnichols102 | had enough of that with FDN this year |
20:25.37 | timscott | bani: you gotta stop talking about pitas like they are a bad thing |
20:25.46 | h3x0r | it would be cheaper to deploy computers and headsets than most voip hard phones |
20:25.52 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
20:25.57 | h3x0r | *cheap* computers anyway |
20:26.00 | Netgeeks | austinnichols, are you looking for termination and origination? |
20:26.05 | austinnichols102 | yup |
20:26.06 | *** join/#asterisk goodjoke (n=Goodjoke@rrcs-24-97-65-74.nys.biz.rr.com) |
20:26.15 | timscott | Hey look, it's a good joke. |
20:26.21 | Netgeeks | do you need any specific DIDs (location wise?) |
20:26.26 | goodjoke | is this room for a@h as well as *? |
20:26.36 | h3x0r | good: read the /topic |
20:26.37 | [TK]D-Fender | [av]bani : No I didn't have atemplate for the one I just set up. like I said, from SCRATCH. |
20:26.43 | goodjoke | thanks..just saw it |
20:26.46 | timscott | oh darn. |
20:26.54 | timscott | Don't leave, it's been a while since I've heard a good joke. |
20:26.56 | timscott | :( |
20:26.59 | Netgeeks | austinnichols102, do you need any specific DIDs (location wise?) |
20:26.59 | timscott | ha ha. :S |
20:26.59 | [TK]D-Fender | [av]bani : I just knew what to look for. |
20:27.21 | goodjoke | i used to have a website called goodjoke.com... then i got married |
20:27.33 | [av]bani | <DTMF tone.dtmf.level="-15" tone.dtmf.onTime="50" tone.dtmf.offTime="50" tone.dtmf.chassis.masking="0" tone.dtmf.stim.pac.offHookOnly="0" tone.dtmf.viaRtp="1" tone.dtmf.rfc2833Control="1" tone.dtmf.rfc2833Payload="101"/> |
20:27.34 | [TK]D-Fender | [16:26] <goodjoke> is this room for a@h as well as *? <- timscott, thats good enough for me :) |
20:27.35 | timscott | it's nice to hear a good joke. |
20:27.38 | [av]bani | hm. that should work, but it doesn't. |
20:27.47 | [av]bani | you punch dtmf and the remote hears nothing. |
20:27.51 | [av]bani | gxp2k's work perfectly. |
20:28.03 | timscott | bani, 'cept for mad lack of echo cancellation |
20:28.11 | [av]bani | timscott: what? |
20:28.15 | timscott | you read me. |
20:28.25 | [av]bani | be more specific? |
20:28.26 | timscott | mad lack of speakerphone echo cancellation |
20:28.32 | [av]bani | er |
20:28.40 | [av]bani | that was fixed in 1.0.2.8 or so |
20:28.43 | [av]bani | last year |
20:28.45 | timscott | Oh, was it? |
20:28.52 | timscott | I updated like four months ago |
20:28.53 | [av]bani | about 8 months ago? |
20:28.55 | h3x0r | http://www.polycom.com/techpartners.htm |
20:28.56 | timscott | i'll update again, I guess. |
20:29.01 | h3x0r | heh they list asterisk business edition |
20:29.04 | [hC] | er? rfc2833 would not let the remote end hear dtmf audibly.. you'd have to use inband for that |
20:29.22 | [av]bani | [hC]: what? |
20:29.31 | [av]bani | [hC]: the gateway converts rfc2833 to audio... |
20:29.36 | h3x0r | and the 600 isnt listed |
20:29.49 | [av]bani | [hC]: if the remote end couldnt hear dtmf audibly, then the gxp2k's wouldnt be working either. but they work. |
20:29.56 | [hC] | [av]bani:: okay, most phones that i use, when using rfc2833, if i press dtmf, talking to someone, they cant hear it |
20:30.02 | [hC] | yet my dtmf works on IVR's and such |
20:30.15 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
20:30.38 | [av]bani | [hC]: um, that's not possible, unless you're talking 100% IP to the ivr |
20:30.47 | [av]bani | if you're going out on pstn, it HAS to be converted to audio. |
20:31.05 | [av]bani | think about it for a minute and i'm sure you'll figure it out :) |
20:31.43 | timscott | yay reboot. |
20:31.50 | timscott | nick war |
20:31.55 | timscott | err |
20:31.59 | timscott | that was the wrong server. |
20:32.38 | [hC] | [av]bani: yes, i agree, it makes sense. Im not sure, i havent really tested what happens extensively. I just know its not an issue :) |
20:33.32 | timscott | hot! |
20:33.36 | timscott | I got my phone to register. |
20:33.37 | timscott | Nice. |
20:33.41 | timscott | Now let us try calling. |
20:34.03 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
20:34.33 | [TK]D-Fender | ok, I'm out, later all |
20:35.46 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-122-123.telkomadsl.co.za) |
20:36.10 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
20:36.26 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
20:36.51 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
20:37.34 | hackeron | Hey, for some reason I hear the occational bleep when going out through ZAP channels (analog lines) and Verizon swear its not them, any ideas what it could be? |
20:39.10 | jsharp | call waiting? |
20:40.23 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
20:40.54 | hackeron | jsharp: hmm, not a feature on the line and isnt it normally a beep? -- I hear like a test blip, its very short and not always clear, just like a blupi type sound, lol |
20:41.16 | docelm0 | Say who in here was building a call center? |
20:41.31 | brif8 | http://pastebin.com/654287 what am I missing ?? |
20:41.42 | jsharp | Dunno, then. |
20:41.54 | lokkju | wondering if someone could give me some suggestions on where to start debugging here. I recently installed * from the debain packages, and am now having trouble with sound - no matter what, I can not seem to get any sound to work. There is no firewall between my client and my server, and there are no error messages in my full log |
20:43.21 | brif8 | getting multiple Registration errors on both servers |
20:45.51 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net) |
20:46.38 | Gamercjm | Has anyone used voipuser.org and set up the DID using SIP? |
20:46.39 | freat | lokkju: you got iptables running on the asterisk server? |
20:46.40 | hackeron | jsharp: I mean there cant really be call waiting, its a hunt group, but maybe it blips when failing over to the next line? - even so, the dialing out lines and incoming lines are at opposite ends |
20:47.00 | Gamercjm | Im having trouble with the SIP URI |
20:47.16 | lokkju | freat, yes, but totally open on the internal interface, which is what I am connecting to |
20:47.37 | freat | lokkju: for grins I would try turning off iptables just to see... |
20:47.53 | lokkju | freat tried it already |
20:47.56 | freat | lokkju: oh ok |
20:48.06 | brif8 | [TK]D-Fender: http://pastebin.com/654287 what am I missing ?? getting multiple registration failure on both servers |
20:48.19 | freat | lokkju: no audio in either direction? if you call a conference room or something with MOH do you hear the music? |
20:49.08 | freat | lokkju: I take it that the phone is registered w/ Asterisk OK? sip show peers |
20:49.46 | freat | also try sip debug and place a call, then sip no debug and scroll up |
20:49.49 | lokkju | yes, and in the debug I can see the CLI saying it is playing sound |
20:50.10 | lokkju | hmm |
20:50.20 | freat | handset plugged in all the way? ;) |
20:50.53 | lokkju | interesting - I put a festival line and a playback line one right after the other, and the festival line I can hear, but not the playback |
20:51.03 | freat | ahh |
20:51.27 | freat | sounds like no timer source |
20:51.46 | freat | check your loaded modules... look for ztdummy |
20:51.51 | *** part/#asterisk goodjoke (n=Goodjoke@rrcs-24-97-65-74.nys.biz.rr.com) |
20:51.53 | freat | or do you have a zaptel card? |
20:51.57 | Dabian | I wonder if you guys could give me some tips on how to set up a PSTN gateway. Both hardware, business model, etc. |
20:52.17 | lokkju | ztdummy *IS* loaded |
20:52.39 | freat | lokkju: hmm.. try re-running ztconfig or whatever the debian equivalent service |
20:52.40 | Dabian | (PSTN/ISDN) |
20:53.07 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
20:53.15 | freat | Dabian: I think people need more specifics. I mean, it could be very simple or very complex depending. |
20:53.15 | Dabian | I assume I will need FXO hardware.. but I also wonder how to scale .. eg. do I need 9 lines for 10 users, or only 5? |
20:53.18 | lokkju | freat, I don't know of any such thing... |
20:53.27 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
20:53.27 | jsharp | Dabian: PRI + Asterisk, buy cheap, sell high. |
20:53.45 | Dabian | freat : Scaleable .. I plan on doing this in my free time. |
20:53.51 | freat | lokkju: there should be a service that loads the ztdummy module |
20:53.59 | Dabian | jsharp : PRI? |
20:54.09 | lokkju | freat, um, loads it into the kernel? |
20:54.55 | freat | well, I dunno about debian. But on linux there's a service that looks for zaptel devices, then if it doesn't find them, it loads up the ztdummy module. |
20:55.15 | jsharp | PRI = Digital phone lines |
20:55.17 | freat | try typing ztcfg at the command line |
20:55.24 | freat | Primary Rate Interface |
20:55.36 | jsharp | 23 lines, brough in on a single circuit. |
20:56.00 | freat | 23 B + 1 D |
20:56.01 | lokkju | no, I just directly load through /etc/modules - and if debian is not linux, then what do you think it is? |
20:56.13 | freat | lokkju: heh yeah I just meant distro |
20:56.15 | Dabian | 23 B? |
20:56.16 | Dabian | jsharp : Sounds nice. How much are they? |
20:56.23 | Dabian | (+/-) |
20:56.38 | freat | $400 month |
20:56.41 | freat | or more |
20:56.47 | freat | depending on your package |
20:56.51 | freat | could be thousands |
20:57.01 | Dabian | Oh .. thats including stuff |
20:57.22 | jsharp | Yah. Price depends on a lot of things. |
20:57.29 | lokkju | freat, got it, all my zt* bins are in my build directory, since I did not want to do a full install - no plans on installing a zaptel card in a 1U in a colo :) |
20:57.44 | jsharp | Then you also have to pay your LD charges on top of the PRI loop charges. |
20:57.58 | Dabian | I would like to route danish PSTN numbers over IP. |
20:58.13 | Dabian | LD? |
20:58.18 | jsharp | long distance |
20:58.23 | b4ka | anyone knows if the sangoma cards work no openbsd with asterisk? |
20:58.32 | freat | Dabian: you should probably read the forums and such |
20:58.36 | Dabian | ahh .. I plan on one way .. PSTN -> IP |
20:59.03 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
20:59.10 | Dabian | At least for starters. Maybe 1 or 2 lines IP->PSTN. |
20:59.29 | Dabian | 23 lines sounds like a nice starting package. |
20:59.36 | jsharp | Oh. Inbound from the PSTN out to IP. |
20:59.36 | Dabian | for PSTN->IP |
20:59.40 | lokkju | freat, ok, so I found the tools - what did you want me to check |
20:59.44 | freat | then just get some regular lines, a digium card w/ 4 ports and go to town |
20:59.52 | freat | lokkju: try running ztcfg |
20:59.57 | jsharp | yeah, what freat said. |
21:00.07 | freat | jsharp: heh |
21:00.09 | lokkju | freat, ok, runs no problems |
21:00.32 | Dabian | jsharp : Yeah .. like someone on PSTN dials "12345678", I pick up the number, look it up in a db or asterisk handles it, and routers it to the SIP client or voicemailbox. |
21:01.45 | jsharp | That's easy, then. |
21:02.07 | Dabian | Heh .. I didn't dare hoping you would say that. :-) |
21:02.28 | jsharp | Its just all a matter of how much you want to spend on hardware and phone line charges. |
21:02.39 | lokkju | wait a sec - this thing about needing usb-uhci still true? |
21:02.45 | Dabian | jsharp : I guess its because I forgot to say I want number-series and numberporting. |
21:02.54 | *** join/#asterisk lyroy (n=toor@modemcable146.87-83-70.mc.videotron.ca) |
21:03.30 | jsharp | Then you'll need a PRI line of some sort, then. |
21:03.41 | lyroy | Does someone here ever experience problems when compiling asterisk-addons on a VIA processor...? |
21:03.41 | websae | has anyone used VoIP-Discount? |
21:04.19 | *** part/#asterisk maffro (n=furby@n156.dkm.cz) |
21:05.35 | lokkju | seems that ztdummy is supposed to rely on usb-uhci, but my system uses uhci_hcd |
21:05.39 | Dabian | jsharp : Eg. lets say Smith has the PSTN number X, and he uses VoIP_Y for IP. Now he wants to go all IP. I tell him, "Pay me some $$, and I'll forward your number." |
21:05.42 | lokkju | could this be causing the issues? |
21:05.47 | *** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com) |
21:06.11 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
21:06.23 | Dabian | jsharp : When someone on PSTN dials X, the number gets on my PRI? and I router it to VoIP_Y. |
21:06.35 | lokkju | (if it makes a diff, I am on the 2.6 kernel) |
21:06.36 | jsharp | Right. Sounds good so far. |
21:06.37 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
21:06.51 | Dabian | jsharp : The PRI, they're not COPPER lines? |
21:06.57 | Dabian | jsharp : They're like VPN? |
21:07.08 | jsharp | They are copper lines, yes. |
21:07.11 | b4ka | anyone knows if the sangoma cards work on openbsd with asterisk? |
21:07.20 | Dabian | jsharp : So I need a bunch of copper? |
21:07.36 | jsharp | Instead of having 24 separate 2-wire phone lines come in, they come in as 24 channels multiplexed together on a T1. |
21:07.47 | jsharp | 4 wires versus 48 wires. |
21:07.53 | jsharp | One port on a switch versus 24 ports. |
21:07.54 | Dabian | ok |
21:08.31 | MikeJ[Laptop] | b4ka, sangoma cards still require zaptel on asterisk |
21:08.42 | MikeJ[Laptop] | so anything that has base zaptel support |
21:09.12 | lyroy | Does someone here ever experience problems when compiling asterisk-addons on a VIA EPIA (mini-itx board with CPU on board)...? |
21:09.14 | Dabian | jsharp: I guess ideally I would prefer that to be handled at the ISP, so I get all the traffic over IP.. |
21:09.34 | jsharp | Where you put the equipment is up to you. |
21:09.47 | Dabian | jsharp : Nice |
21:09.48 | jsharp | But if you put it in a colocation facility, you often get a much better price on the PRI. |
21:10.04 | Dabian | jsharp : Ahh of course.. Colocation! |
21:10.41 | Dabian | jsharp : That way I might not have to pay for digging down copper (which would probably make it virtually impossible for me) |
21:11.07 | *** join/#asterisk fr3aky (n=fr3aky@nar.macol.ru) |
21:12.13 | *** join/#asterisk dwmw2 (n=dwmw2@baythorne.infradead.org) |
21:15.03 | Dabian | jsharp : I don't really want to offer outgoing PSTN - at least not for starters, since I assume thats where you put the heavy money down. |
21:15.47 | shido6 | anyone in India? |
21:15.52 | freat | lokkju: sorry got a phone call |
21:16.12 | freat | lokkju: can you place outbound or inbound calls? |
21:16.34 | Dabian | jsharp : The idea is to make it easy for people to switch between different VoIP companies offering peering, pertaining their old PSTN number for a small fee they pay me. |
21:17.23 | Dabian | jsharp : They wont get CID on the number though. |
21:18.16 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
21:18.36 | Dabian | jsharp : I am trying to figure out if its possible to do within a buget. I don't want to rip off people, on the other hand, I don't want to work for free. |
21:19.14 | Dabian | !ping |
21:19.18 | lyroy | Does someone here ever experience problems when compiling asterisk-addons on a VIA EPIA (mini-itx board with CPU on board)...I have some message like that ( mysql.h No such file...) ? |
21:19.48 | Dabian | jsharp : Any idea about porting numbers, and number series? |
21:20.42 | Dabian | jsharp : I mean, every time I get a number ported, I guess I have to tell the PRI provider, "Hey, would also route the number XXXXXXX for me?" |
21:21.20 | lokkju | freat, inbound, no problem |
21:21.41 | lokkju | freat, call <anything>@ns2.ifpdx.com |
21:22.09 | lokkju | you are supposed to hear a festival voice, then three beeps |
21:22.14 | Dabian | freat: How much is a digium 4 ports card, btw? (Not exact price, just the ball park) |
21:24.11 | Nugget | http://store.digium.com/ |
21:24.18 | lokkju | Dabian, http://www.gmprice.com/index.php?qstring=digium+4&cat=61838 |
21:24.44 | lokkju | average ebay price is $342, as you can see on the site I just pasted |
21:24.59 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
21:25.00 | lokkju | (my own little tool, and yes, I'll plug it when I can) |
21:25.13 | websae | anyone here manage or own a call center, where you use only SIP or IAX trunks for termination? How is your call quality and uptime? |
21:26.34 | websae | i am curious how many businesses are going to VoIP trunks either via SIP or IAX |
21:27.26 | lokkju | freat, please, some more ideas on debugging this |
21:29.57 | brodiem | Is it possible to dial out in a Meetme room? |
21:30.13 | badboyz | so if i have a t1, and i want to connect it to an asterisk box, to use as a backup for outgoing phone calls... which interface card do i need? |
21:32.03 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
21:32.22 | PakiPenguin | is there a difference between an integrated t1 and mixed mode t1 |
21:32.35 | Dabian | websae : Maybe that would be interesting for me.. |
21:32.45 | Dabian | lokkju : Sounds kinda expensive. |
21:32.53 | Dabian | lokkju : I assume its high quality? |
21:34.40 | *** join/#asterisk stoffell (n=stoffell@pot.catsanddogs.com) |
21:35.01 | lokkju | Dabian, they are digium *shrug* - and that actually pretty damn cheap compared to normal phone hardware |
21:35.50 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
21:36.55 | *** join/#asterisk nagl (n=nagl@86.59.54.238) |
21:37.44 | Dabian | Oh .. I don't mean expensive .. even though thats what I wrote .. just more pricey than I expected. |
21:38.31 | Nivex | brodiem: not directly. You could use the manager interface or a call file to bring a party in though |
21:39.18 | skyboy | hello, can anyone help out with a config regarding ser to asterisk connectivity? |
21:40.10 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
21:40.45 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
21:41.02 | Renacor | Is there an app that can take the asterisk cdr csv and parse it for statistics? |
21:41.38 | Aurs | Renacor: have you considered cdr_odbc or cdr_mysql? |
21:43.24 | Renacor | is there a howto on it? |
21:43.33 | Renacor | Im guessing you can make cdr_odbc work with postgres? |
21:44.25 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
21:45.01 | PakiPenguin | apple.com :) |
21:45.04 | PakiPenguin | hehe |
21:45.07 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
21:45.10 | PakiPenguin | send me a free imac |
21:45.12 | PakiPenguin | please |
21:45.21 | tainted- | skyboy what are u trying to do? |
21:46.56 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
21:46.59 | skyboy | tainted: I am actually trying to track down a weird situation in SER config to proxy to asterisk. Ive come to understand the "vernacular" used in ser but doesnt understand the resolution bits. |
21:47.05 | Aurs | Renacor: don't know if there is a howto... but if there is, google will know ;) |
21:47.10 | skyboy | here is an example - |
21:47.27 | Dabian | What makes digum cards pricey? What can they do, that your old soundcard with FXO wont do? |
21:47.46 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
21:48.15 | Nugget | work without you having to write your own drivers. |
21:48.21 | eric_hill | Can anyone tell me how to automatically log off a dynamic ACD agent if they don't answer? The automatic logoff function only seems to work for agents defined in agents.conf... |
21:48.23 | Dabian | Do they encode the voice in the right codec? |
21:48.55 | skyboy | if (uri=~"^sip:\+[0-9]+@192.168.0.1") { |
21:48.55 | skyboy | <PROTECTED> |
21:48.55 | skyboy | rewriteport(""); |
21:48.55 | skyboy | <PROTECTED> |
21:48.55 | skyboy | <PROTECTED> |
21:49.01 | skyboy | sorry about that... |
21:49.07 | Dabian | Nugget : Ahh ok. Thats of course an interesting point. Writing drivers can be quite timeconsuming. :-) |
21:49.11 | skyboy | but thre it is... |
21:50.00 | skyboy | tainted: basically the there is a forward im guessing to a machine known as pbx.foo.org but when I do a dns lookup there is no such machine..weird. |
21:50.10 | skyboy | am I reading the config file correctly? |
21:51.37 | Renacor | hmm i got the cdr in a postgres database, however I don't think there is any way to get call statistics for agents answering a queue from the data in the cdr is there? |
21:52.48 | skyboy | any ideas |
21:54.43 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
21:54.45 | DeeJay[2] | Hi! |
21:54.55 | DeeJay[2] | Has anybody been able to use BLF with polycom phones? |
21:55.40 | Katty | hihi. |
21:55.42 | triple-e | whats BLF ? |
21:55.53 | Qwell[] | ~blf |
21:55.55 | jbot | extra, extra, read all about it, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
21:56.34 | Dabian | jsharp : Thanks for the advise .. I will look into the PRIs. |
21:59.02 | websae | does anyone here have experience with ASTBILL? |
22:00.01 | lokkju | wtf is up with this - no sound from Playback, using ztdummy - everything looks ok, but it as if it has no timer source - any way to actually test that ztdummy is fully working? |
22:11.15 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
22:11.51 | *** join/#asterisk Saturn-- (i=Saturn@24.50.85.195) |
22:13.51 | Dabian | Is it true that asterisk doesn't support crypto-sip? |
22:14.09 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
22:14.15 | MattB2 | hi all |
22:14.17 | MattB2 | got a qq |
22:14.41 | MattB2 | if someone calls my asterisk and in the dialplan i set a variable, will that variable already be set when another user calls in? |
22:14.47 | MattB2 | or are they per-call |
22:18.28 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
22:19.38 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
22:21.45 | PakiPenguin | tzafrir, around? |
22:22.05 | Renacor | anybody know an app that can work with a postgres cdr and create statistics for answered calls by agents? |
22:22.20 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
22:22.21 | Gamercjm | Im having problem with SIP to SIP, like the users cannot hear each other |
22:22.44 | websae | SIP to SIP is terrible |
22:22.47 | websae | because of RTP |
22:22.50 | websae | need to use IAX :) |
22:23.02 | Gamercjm | yah but i need to use sip |
22:23.10 | Gamercjm | i have the RTP ports open on the server |
22:24.26 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
22:25.10 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
22:26.20 | mitcheloc | Renacor: snap snap snap ;) (well not yet, but it will) |
22:27.06 | lokkju | Renacor, would be easy to do - just write a couple sql queries, and a php page to display em |
22:28.57 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
22:30.40 | *** join/#asterisk Maxxed (n=user@cpe-72-177-150-20.houston.res.rr.com) |
22:30.45 | Maxxed | yo fellas :) |
22:31.42 | Maxxed | gota odd one |
22:31.42 | Maxxed | > reload chan_zap.so |
22:31.42 | Maxxed | -- Reloading module 'chan_zap.so' (Zapata Telephony) |
22:31.42 | Maxxed | == Parsing '/etc/asterisk/zapata.conf': Found |
22:31.42 | Maxxed | Apr 11 16:31:14 ERROR[4284]: chan_zap.c:10309 setup_zap: Unable to reconfigure channel '1' |
22:31.42 | Maxxed | Apr 11 16:31:14 WARNING[4284]: chan_zap.c:11069 reload: Reload of chan_zap.so is unsuccessful! |
22:31.54 | Maxxed | and idea? |
22:31.58 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-174-212.lsanca.fios.verizon.net) |
22:31.59 | Maxxed | my cfgs look stright |
22:32.03 | Maxxed | mods look loaded |
22:32.10 | Maxxed | ztcfg is showing sweet |
22:32.38 | Maxxed | tdp400p |
22:32.44 | Maxxed | two fxo ifaces |
22:32.50 | Maxxed | using fxsks signaling |
22:33.19 | Maxxed | im about to recompile |
22:35.31 | key2 | someone has an idea of why caller ID on X100P doesn't work ? |
22:35.49 | Maxxed | im not sure, iv never used the x100p's |
22:35.54 | Maxxed | it should suport it if im not mistaken |
22:36.23 | Maxxed | check ye ol zapta.conf |
22:36.33 | Maxxed | usecallerid=yes |
22:36.47 | key2 | yeah |
22:37.02 | Maxxed | do you have caller id service? |
22:37.14 | Maxxed | i have to pay a few bux extra a month for it on my analog crap |
22:37.25 | key2 | yeah |
22:37.32 | key2 | since with a phone I see the number |
22:37.47 | key2 | so I do have |
22:37.53 | Maxxed | are you trying to set the outgoing caller id ? |
22:37.58 | Maxxed | or just view the incoming |
22:37.58 | key2 | no |
22:38.05 | key2 | view incomming |
22:38.07 | Maxxed | um |
22:38.15 | key2 | it's not possible on analog line to set outgoing callerid |
22:38.24 | Maxxed | ok, so you know that much ;) |
22:38.35 | Maxxed | ive seen alota folks come thru that didnt know that |
22:38.36 | Maxxed | heh |
22:38.36 | key2 | lol |
22:38.42 | Maxxed | lol n'deed |
22:39.05 | key2 | is it possible to be a Line/Phone issue ? |
22:39.55 | key2 | I tryed with both actually and it's the same :) |
22:39.56 | Maxxed | it may be.. do you have a caller id u can plug in |
22:39.56 | Maxxed | to test if your id service is working |
22:39.56 | skyboy | asked on the ser forum..but maybe someone can help me here ---how do you query a dns server for srv records? |
22:39.56 | key2 | Maxxed: yeah, I tryed with an analog phone |
22:39.56 | Maxxed | or a plane ol phone that supports callerid |
22:40.03 | *** join/#asterisk CpuID2 (n=none@gentoo/contributor/cpuid) |
22:40.08 | key2 | and it gives me the id |
22:40.08 | Maxxed | skyboy: are you familair with dig? |
22:40.24 | Maxxed | key2: um, sounds like a config some where |
22:40.40 | Maxxed | key2: cuz im pretty sure the xp100p's have caller id support |
22:40.42 | skyboy | Maxxed: familiar in that I have used to do a dig against a dns server.. |
22:41.38 | skyboy | Maxxed: but im perplexed because I have record that forwards in SER to pbx.foo.org but doing an nslookup against that resolves to nothing...so im thing where is that machine? |
22:42.05 | Maxxed | key2: dig @dns1.whateversrever.com domain.net any |
22:42.08 | skyboy | Maxxed: Am I thinking of it wrong? |
22:42.09 | Maxxed | key2: i think.. ;p |
22:42.37 | Maxxed | wops |
22:42.38 | Maxxed | wrong nicks ;p |
22:42.38 | skyboy | n/p |
22:42.38 | skyboy | I got it ;) |
22:42.39 | Maxxed | rockin :) |
22:42.46 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-174-51.lsanca.fios.verizon.net) |
22:43.00 | Gamercjm | ok so im still having a sip to sip audio problem |
22:43.02 | skyboy | but ill have to check if it resolves back :) |
22:43.07 | Gamercjm | anyone know how to get that working |
22:43.39 | Maxxed | key2: go over the configs, check logs, what kinda phones are you using? |
22:44.05 | Maxxed | key2: might have some wacky extentions.conf messin it up |
22:44.13 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
22:44.56 | Maxxed | anyone have any idea why im geting this " setup_zap: Unable to reconfigure channel '1' " crap ? |
22:45.12 | Maxxed | im about to recompile, maybe somthing gone south the frist go around |
22:45.36 | Maxxed | asterisk seems to be working well, minus the tdm400p |
22:45.49 | Az_au | did the kernel modules cause any errrors? |
22:45.54 | Maxxed | nope |
22:45.57 | Maxxed | they loaded fine |
22:46.02 | Az_au | try rmmod and modprobe again? |
22:46.21 | Az_au | if not i'd recompile both zaptel and asterisk |
22:46.31 | Maxxed | yeah.. |
22:46.34 | Maxxed | Apr 11 15:29:55 host kernel: Zapata Telephony Interface Registered on major 196 |
22:46.34 | Maxxed | Apr 11 15:29:55 host kernel: Zaptel Version: 1.2.4 Echo Canceller: KB1 |
22:46.34 | Maxxed | Apr 11 15:29:55 host kernel: ACPI: PCI interrupt 0000:01:01.0[A] -> GSI 16 (level, low) -> IRQ 193 |
22:46.34 | Maxxed | Apr 11 15:29:55 host kernel: Freshmaker version: 71 |
22:46.34 | Maxxed | Apr 11 15:29:55 host kernel: Freshmaker passed register test |
22:46.35 | Maxxed | Apr 11 15:29:55 host kernel: Module 0: Installed -- AUTO FXO (FCC mode) |
22:46.37 | Maxxed | Apr 11 15:29:55 host kernel: Module 1: Installed -- AUTO FXO (FCC mode) |
22:46.39 | Maxxed | Apr 11 15:29:55 host kernel: Module 2: Not installed |
22:46.41 | Maxxed | Apr 11 15:29:55 host kernel: Module 3: Not installed |
22:46.43 | Maxxed | Apr 11 15:29:55 host kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules) |
22:46.45 | Maxxed | Apr 11 15:29:55 host kernel: Registered tone zone 0 (United States / North America) |
22:46.49 | Maxxed | thx for the pointers Az_au |
22:46.50 | eric_hill | Can anyone tell me how to automatically log off a dynamic ACD agent if they don't answer? The automatic logoff function only seems to work for agents defined in agents.conf... |
22:47.04 | skyboy | Maxxed: it comes back with the authoritative dns server but no record per say |
22:47.09 | Maxxed | # lsmod |
22:47.09 | Maxxed | wctdm 40192 0 |
22:47.10 | Maxxed | zaptel 228772 5 wctdm |
22:47.26 | Maxxed | skyboy: try dig @dns1.whateversrever.com domain.net ser |
22:47.34 | skyboy | okay |
22:47.50 | Maxxed | dnstools.com might have an easy point click way to find out |
22:48.02 | Maxxed | we i mean dnsstuff |
22:48.02 | Maxxed | http://www.dnsstuff.com/ |
22:48.06 | Az_au | Maxxed: i've got 3 modules loaded (i'm using fxs and fxo) |
22:48.07 | Az_au | wcfxo 17312 0 |
22:48.07 | Az_au | wctdm 41920 2 |
22:48.07 | Az_au | zaptel 192516 10 wcfxo,wctdm |
22:48.07 | Maxxed | s/we/er |
22:48.08 | Maxxed | heh |
22:48.24 | Maxxed | um, i dont have the 3rd |
22:48.27 | Maxxed | wcfxo ? |
22:48.34 | Az_au | i think it's an alias |
22:48.35 | Maxxed | you have a tdm400p ? |
22:48.36 | Az_au | lemme check |
22:48.37 | Az_au | ya |
22:49.02 | Maxxed | the readme states, wctdm or wcfxs |
22:49.05 | Maxxed | i think.. |
22:49.15 | *** join/#asterisk GTX (n=charlie@pdpc/supporter/monthlybronze/GTX) |
22:49.17 | Az_au | alias wcfxs wctdm |
22:49.22 | GTX | Guys, I have an snom 360 and in the SIP settings it says "Music On Hold Server" does anyone know one of these? |
22:49.23 | Maxxed | wctdm or TDM400P - Modular FXS/FXO interface (1-4 ports) |
22:49.23 | Maxxed | wcfxs |
22:49.29 | Az_au | so my fxs is covered by wctdm but fxo seems to be it's own |
22:49.45 | Az_au | my modprobe.conf lists as follows for fxo: |
22:49.46 | Az_au | install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg |
22:49.47 | Maxxed | i have two fxo modules on the tdm400p |
22:49.51 | skyboy | Maxxed: okay...the dig im doing is supposed to be asterisk box so does the prvious dig command still hold? |
22:50.02 | skyboy | Here is the config - |
22:50.44 | skyboy | if (uri=~"^sip:\+[0-9]+@someip") { |
22:50.44 | skyboy | <PROTECTED> |
22:50.44 | skyboy | rewriteport(""); |
22:50.44 | skyboy | <PROTECTED> |
22:50.44 | skyboy | <PROTECTED> |
22:50.56 | Maxxed | install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg |
22:51.19 | skyboy | Maxxed: the pbx.foo.org appears to me where SER is forwarding correct? |
22:51.22 | Maxxed | alias wcfxs wctdm |
22:51.44 | *** join/#asterisk Op3r (n=op3r@203.82.42.10) |
22:51.46 | Maxxed | skyboy: um.. not sure, i havent toyed with any dns stuff in a long while ;\ |
22:51.51 | *** join/#asterisk melange8272 (n=melange8@ool-4576ab1f.dyn.optonline.net) |
22:53.02 | Az_au | maxed: my startup modprobe goes like this: |
22:53.05 | Az_au | modprobe zaptel |
22:53.05 | Az_au | modprobe wcfxs |
22:53.05 | Az_au | modprobe wcfxo |
22:53.29 | Maxxed | i dont have any fxs modules, so i wouldnt think i would need it |
22:53.49 | Az_au | nah proll ynot |
22:54.01 | Maxxed | im gona try and recompilke |
22:54.05 | Maxxed | blah |
22:54.07 | Maxxed | recompile |
22:54.46 | Maxxed | i do remember having a few small issues with the zaptel compile. dependcy issues i think |
22:54.59 | Maxxed | il swing back by and harass you guys if i need anything |
22:55.03 | Maxxed | thanks again Az_au |
22:55.06 | Az_au | np |
22:55.25 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
22:55.52 | *** part/#asterisk GTX (n=charlie@pdpc/supporter/monthlybronze/GTX) |
22:57.13 | Gamercjm | Is there anyway to get the rtp thing in SIP to not have to open rtp ports on the client router? |
23:00.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
23:00.22 | Sedorox | exit |
23:00.22 | Sedorox | exit |
23:00.24 | Sedorox | errrrr |
23:04.16 | Gamercjm | How do i add a SDP thing |
23:08.29 | melange8272 | anyone know if sip digits are spit out over the wire in call (sniffable?) I know the invite is easy to snag, but not sure if digits are sent rtp during the call |
23:10.51 | Maxxed | well hell ;p |
23:11.03 | Maxxed | Az_au: i recompiled the zzaptel drivers, they work like a champ now |
23:11.04 | Maxxed | heh |
23:11.07 | Maxxed | how bout that ;) |
23:11.18 | Qwell[] | melange8272: yes, dtmf is in the rtp |
23:12.03 | melange8272 | sniffable in ethereal.. Guess I won't check my bank balance from my sip phone ;) |
23:15.20 | *** join/#asterisk timscott (n=a@d198-166-221-177.abhsia.telus.net) |
23:15.23 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
23:15.25 | gbodemantv | hi all |
23:15.31 | gbodemantv | so who is using xlite? |
23:17.11 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
23:17.37 | tainted- | they're all trying to figure out what's wrong with it right now |
23:17.49 | MRH2 | hi can anyone tell what : #define MONITOR_CONSTANT_DELAY does in channel.c |
23:18.10 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
23:19.30 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
23:19.58 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
23:20.47 | MoutaPT | Hi, does any one knows or could guess why Xlite is taking like 5 or 6 seconds to Hung up a call, ASterisk is 1.2.5? |
23:24.00 | FLeiXiuS | lol to hung it up? |
23:24.05 | FLeiXiuS | Hang up a call maybe? ;_) |
23:25.37 | MoutaPT | you r right... thks |
23:25.46 | MoutaPT | :) my english isn't so good |
23:26.05 | MoutaPT | anyways you got the question?:) |
23:26.15 | MRH2 | is there some emergency going on? |
23:26.59 | MRH2 | this channel seems a bit lite at the moment |
23:26.59 | MoutaPT | MRH2 what kind of emergency? not just a normal call... |
23:27.13 | MoutaPT | yeah u r right MRH2 |
23:28.48 | gbodemantv | trying to turn off call waiting sound in xlite |
23:28.53 | gbodemantv | anybody have any idea how |
23:28.54 | gbodemantv | ?? |
23:31.25 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
23:32.40 | *** join/#asterisk mrbnet (n=sureal@cust-static194-37.BHI.COM) |
23:33.46 | mrbnet | has anyone here setup a DID using virtualphoneline.com that could answer a couple q's for me? |
23:38.06 | ljam | how might one go about logging an agent out of a queue? |
23:43.16 | [av]bani | http://siln.livejournal.com/235478.html |
23:44.41 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
23:45.53 | brif8 | I have IP Phone <----> * Box 1 <---> Gateway (NAT) <---> Internet <---> Gateway Nat <---> * Box 2 <---> T1 Card. How can I route calls from the T1 Card on Box 2 to IP Phones on Box 1 ? |
23:47.52 | tzanger | werd to the goat herd |
23:48.06 | timscott | brif: over the internet |
23:48.07 | timscott | :p |
23:48.28 | timscott | asterisk, man. install and set it up. |
23:50.04 | skyboy | hello..im incurring some errors when bulding zaptel-1.2.5 on centos4 --> errer:syntax error before "zone_lock" can someone help out? |
23:50.14 | skyboy | then followed by a few more ;) |
23:51.09 | skyboy | I set up the simlinks before the make properly so Im perplexed as to what package Im missing..or?? |
23:51.45 | X-Rob | skyboy, |
23:51.50 | X-Rob | ~centosbug |
23:51.51 | jbot | from memory, centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
23:52.23 | skyboy | jbot: thanks... |
23:52.23 | jbot | skyboy: sure thing |
23:52.48 | brif8 | timscott: I'm trying to connect via IAX2 but and I read the wiki on dual servers but for some reason when I dial the exten on Box 2 to ring a phone on Box 1 it rings once and then disconnects ?? |
23:52.52 | *** join/#asterisk wgroh (n=chatzill@69-163-232-176.atlsfl.adelphia.net) |
23:53.20 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
23:53.37 | skyboy | jnot: one more thing am I playing with "fire" with gas on my hands ;) by using 4.3 or shoud the rest of the install go smoothly. In particular what is the recommended version to go with for stability.. |
23:53.48 | X-Rob | skyboy, jbot is a bot. |
23:53.53 | brif8 | I was hoping someone might be able to assist what dumb mistake I am making in the config or something. |
23:54.16 | *** join/#asterisk rva (n=rafa@200.210.51.130) |
23:54.17 | X-Rob | brif8, usually codec incompatibilites. look in /var/log/asterisk/full |
23:54.23 | skyboy | ahh...learning something new everyday ;) |
23:54.38 | rva | hi guys....could someone give me a little help with realtime? |
23:54.59 | rva | asterisk seems not to be sending the user authentication to the realtime....mysql |
23:55.20 | skyboy | hmm...is 4.3 version of centos stable for * or should I be using a previous version for stability? |
23:55.22 | brif8 | X-Rob: there is no /var/log/asterisk/full |
23:55.32 | brif8 | events and messages |
23:55.37 | X-Rob | brif8, edit /etc/asterisk/logger.conf and turn it ou |
23:55.50 | wgroh | anyone having DTMF problems with the TE406 |
23:55.51 | X-Rob | skyboy, centos4.3 is heavily used |
23:55.54 | wgroh | the VPM |
23:56.18 | skyboy | sweet |
23:56.28 | skyboy | X-Rob: thanks |
23:56.40 | brif8 | messages => error,warning and console => error,notice,warning;,info |
23:58.41 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
23:58.55 | brif8 | X-Rob: I see Unable to create translator path for unknown to slin on IAX2/lecanto-16384 in messages |
23:59.04 | wgroh | looks like the TE420 is the same as TE406 but with hardware DTMF |
23:59.14 | Qwell[] | hardware dtmf? |
23:59.23 | Qwell[] | I think you mean echo cancel... |
23:59.27 | tzanger | no |
23:59.28 | wgroh | negative |
23:59.37 | tzanger | hardware DTMF detection and other little goodies |
23:59.38 | Qwell[] | does the TE420 even exist? |
23:59.40 | brif8 | also chan_iax2.c: Max retries exceeded to host 71.41.50.162 on IAX2/lecanto-16384 (type = 6, |
23:59.42 | wgroh | chan_zap.c: Detected digit 'A' |
23:59.45 | wgroh | chan_zap.c: Detected digit 'D' |
23:59.57 | wgroh | TE406 is killing my team |