00:00.00 | salimfadhley | If they are using iax2 without a secret that means anybody could pretend to be me... nice. |
00:01.40 | SplasPood | http://www.protestwarrior.com/nimages/the_sign.jpg |
00:01.41 | SplasPood | heh |
00:04.38 | redder86 | is there a way to query if an extension is in-use? setgroup/checkgroup doesn't perfectly fit the bill for me because I often dial multiple extensions like this Dial(extension1&extension2), and setgroup is useless here in distinguishing which extension answered the call. |
00:06.45 | visik7 | anyone use res_config? |
00:07.39 | bkw_ | salimfadhley you didn't fix the problem |
00:07.43 | bkw_ | visik7 yes why? |
00:08.02 | bkw_ | redder86 doesn't matter it will still work |
00:08.24 | bkw_ | you want to know who answered you're SOL in that case |
00:08.46 | bkw_ | thats why blah&blah2&blah3 suck on dial |
00:09.02 | visik7 | bkw_ I can't figure how it works wiki page is not so clarifying |
00:09.57 | *** join/#asterisk habakuk (~chatzilla@adsl-64-168-23-239.dsl.snfc21.pacbell.net) |
00:11.01 | twisted | hahaha |
00:11.05 | twisted | "we're gonna flex the tight end" |
00:13.13 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
00:13.29 | *** join/#asterisk doughecka_ (~dheckaman@doughecka.user) |
00:13.30 | bkw_ | visik7 res_config in cvs-head is diffrent |
00:13.31 | doughecka_ | Running ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
00:13.35 | bkw_ | no device |
00:13.37 | bkw_ | reinstall drivers |
00:13.49 | bkw_ | or fix zaptel.conf |
00:14.03 | doughecka_ | only happens after kudzu runs |
00:14.07 | doughecka_ | and I turned it off |
00:14.11 | doughecka_ | but it still happened |
00:14.43 | doughecka_ | same error |
00:14.56 | ManxPower | doughecka: I'll bet kudzu thinks an ixj is in the system. |
00:15.23 | ManxPower | remove the card, run kudzu to remove the configs, put the card back in, tell kudzu to not install the card |
00:16.08 | twisted | heh |
00:16.10 | twisted | better yet |
00:16.13 | twisted | remove kudzu |
00:16.36 | *** join/#asterisk gabe (~gabe@c-064-186-245-122.sd2.redwire.net) |
00:16.41 | PoWeRKiLL | Any idea why I just restart my sever and asterisk is running back fine but when I call to moh or voicemail it's not working anymore ? |
00:16.58 | redder86 | bkw_: is there any option other than Dial(blah&blah2&blah3) ? |
00:17.27 | doughecka_ | twisted: heh |
00:17.35 | JunK-Y | redder86: what ya wanna do exactly ? |
00:17.45 | bkw_ | redder86 not really |
00:18.06 | redder86 | JunK-Y: I want to only have call-waiting work to the extensions for specified callers |
00:18.15 | PoWeRKiLL | The only thing i've done is remove kernel support of RTC and compile zaptelrtc |
00:18.25 | PoWeRKiLL | do you think this is the pb ? |
00:18.56 | bkw_ | ya think thats the issue.. gee I wonder |
00:19.15 | redder86 | JunK-Y: I don't want to have my ongoing calls disturbed by other people unless they are in my "priority calls" list. But I *do* want to have those priority callers get through via call-waiting. |
00:19.17 | doughecka_ | still giving me the error |
00:19.56 | ManxPower | doughecka: lsmod shows the ixj module loaded? |
00:20.06 | doughecka_ | ManxPower: its rebooting now |
00:20.10 | redder86 | JunK-Y: so I can't disable call-waiting on the phones. I need to have a way to see if the extension is in-use before Dialing it |
00:20.13 | Damin | No. |
00:20.24 | tzanger | doughecka_: can you help me with firefly? |
00:20.29 | doughecka_ | possibly |
00:20.34 | ManxPower | redder86: See "setGroup" in the wiki |
00:20.36 | doughecka_ | its mindboglinly simple |
00:20.41 | tzanger | doughecka_: I thought so too |
00:20.43 | doughecka_ | mindboggingly |
00:20.46 | tzanger | I can dial |
00:20.49 | doughecka_ | ooh |
00:20.51 | doughecka_ | recieving calls |
00:20.54 | JunK-Y | redder86: and its to dont disturb them right ? |
00:20.56 | tzanger | it gives me a funky ringback though (sounds european) |
00:20.56 | doughecka_ | makesure callerid is passed |
00:20.58 | tzanger | and it connects |
00:21.01 | tzanger | but no audio is passed |
00:21.03 | doughecka_ | oh |
00:21.04 | tzanger | just silence in both directions |
00:21.05 | doughecka_ | no audio? |
00:21.11 | doughecka_ | make sure your codecs match |
00:21.12 | ManxPower | doughecka: So that fixed it? |
00:21.13 | redder86 | ManxPower: setGroup doesn't work exactly as I would need it when I dial extensions with multiple entries: Dial(x&y&z) |
00:21.20 | tzanger | doughecka_: they should match |
00:21.20 | tzanger | heh |
00:21.21 | redder86 | JunK-Y: yes |
00:21.30 | ManxPower | redder86: Use chan Local. also see the Wiki |
00:21.33 | doughecka_ | ManxPower: no, lsmod? |
00:21.44 | ManxPower | lsmod lists the modules loaded in Linux |
00:21.44 | doughecka_ | # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
00:21.47 | doughecka_ | eep |
00:21.51 | doughecka_ | # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
00:21.55 | ManxPower | doughecka: But it it's working..... |
00:21.55 | doughecka_ | blah, stupid copy and paste |
00:22.00 | JunK-Y | redder86: that could be interesting yes, ive DND button on my phones, if ya find a way, please let me know how, im curious. |
00:22.03 | PoWeRKiLL | anyone using zaptelrtc ? |
00:22.14 | doughecka_ | ManxPower: asterisk does not start up |
00:22.21 | doughecka_ | ztcfg -vv gives that error |
00:22.41 | doughecka_ | ok |
00:22.47 | doughecka_ | it shows zaptel loaded |
00:22.49 | *** join/#asterisk Phantom-X (someone@h133n2fls24o1068.bredband.comhem.se) |
00:22.51 | doughecka_ | used by wcusb |
00:22.55 | *** join/#asterisk VoiceLynx (~rda@user-0cdv656.cable.mindspring.com) |
00:22.55 | Phantom-X | hello there |
00:22.58 | doughecka_ | wcusb is unused and loaded |
00:24.01 | doughecka_ | wcfxo says no such device |
00:24.05 | doughecka_ | modprobe wcfxo * |
00:24.23 | tzanger | doughecka_: |
00:24.24 | tzanger | <PROTECTED> |
00:24.24 | tzanger | <PROTECTED> |
00:24.24 | tzanger | <PROTECTED> |
00:24.24 | tzanger | <PROTECTED> |
00:24.26 | tzanger | <PROTECTED> |
00:24.27 | JunK-Y | ~seen lennyt |
00:24.32 | jbot | lennyt <~lenny@rockbox-gw.voiping.com> was last seen on IRC in channel #asterisk, 5d 6h 36m 16s ago, saying: 'since the pap2-na is not avail ... anyone figure how to unlock the pap2?'. |
00:24.32 | tzanger | <PROTECTED> |
00:24.32 | tzanger | looks like it would work to me |
00:24.34 | tzanger | but no audio |
00:24.53 | doughecka_ | tell it to disallow=all allow=ilbc |
00:25.01 | doughecka_ | and turn ilbc on in firefly |
00:25.24 | PoWeRKiLL | ~seen kapejod |
00:25.25 | jbot | kapejod <~kapejod@pD9E83D65.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 97d 8h 4m 29s ago, saying: 'later on you will be able to use any elcheapo hfc-pci card'. |
00:25.34 | Phantom-X | im more or less unfamliliar with asterisk , I joined here just to get some info , I cot a cisco ata 186 , right now its wunning behind a NAT:ing firewall , now I wonder if I could use asterisk to act as proxy/answering machine etc I mean when im not at home I wnant somethging to answer the dial or redirect it |
00:26.38 | ChulJin | yes. |
00:26.43 | *** join/#asterisk crdobbs (~crdobbs@175.247.232.64.transedge.com) |
00:26.48 | Phantom-X | is it hard to achieve it ? |
00:26.55 | doughecka_ | ManxPower: still having the problem |
00:27.14 | doughecka_ | removed the hardware entry in /etc/sysconfig/hwconf |
00:27.17 | doughecka_ | and rebooted |
00:27.22 | doughecka_ | still gives the same error |
00:27.25 | tzanger | ew ilbc |
00:27.25 | doughecka_ | cant find the card.. |
00:27.28 | doughecka_ | tzanger: ew? |
00:27.31 | ManxPower | doughecka: What about the lsmod showing? |
00:27.31 | doughecka_ | then try ulaw |
00:27.36 | ChulJin | ~ManxPower's 'Useful Asterisk Docs' |
00:27.43 | doughecka_ | ~manxpower |
00:27.45 | jbot | Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section also see http://www.voip-info.org/wiki-Asterisk also see http://www.fnords.org/~eric/asterisk/ |
00:27.46 | ChulJin | hmmm |
00:27.48 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
00:27.57 | doughecka_ | ManxPower: heh, zaptel and wcusb are loaded |
00:28.09 | doughecka_ | wcfxo is NOT loaded |
00:28.09 | ManxPower | doughecka: ixj isn't loaded? |
00:28.16 | doughecka_ | ixj? |
00:28.16 | ChulJin | dough: hehe I knew jbot would have to have memorised those by now. |
00:28.17 | doughecka_ | hmm, no |
00:28.19 | ManxPower | doughecka: Is this a new card or a new system? |
00:28.24 | doughecka_ | both |
00:28.26 | ManxPower | and if you modprobe wcfxo ? |
00:28.29 | doughecka_ | just installed OS |
00:28.36 | doughecka_ | that says it cant find the device |
00:28.39 | doughecka_ | this is a TDM400 |
00:28.44 | doughecka_ | with 4 FXO |
00:28.48 | ManxPower | You need wcfxs |
00:28.59 | tzanger | doughecka_: interesting -- it still negotiates GSM even though iax.conf has disallow=all/allow=ilbc and firefly only has ilbc checked |
00:29.05 | tzanger | reloaded * and exited/reloaded firefly |
00:29.07 | doughecka_ | tzanger: reload? :) |
00:29.07 | ManxPower | doughecka: the zaptel README lists the module -> card info |
00:29.12 | Phantom-X | thanks for the urls |
00:29.12 | doughecka_ | shoot :) |
00:29.26 | doughecka_ | ManxPower: oh, I try loading the right module |
00:29.37 | ManxPower | doughecka: It usually makes things work better 8-) |
00:29.40 | doughecka_ | heh |
00:29.50 | ManxPower | citats: Are you around? |
00:30.21 | doughecka_ | or, hopefully will be |
00:30.23 | ManxPower | doughecka: You like pain, don't you? |
00:30.26 | doughecka_ | well |
00:30.27 | tzanger | doughecka_: hmm reSTARTed asterisk and it still negotiates GSM |
00:30.27 | doughecka_ | it WAS working |
00:30.29 | doughecka_ | great |
00:30.31 | doughecka_ | till I rebooted |
00:30.42 | doughecka_ | tzanger: sux to be you :) |
00:30.50 | doughecka_ | thirdparty version correct? |
00:31.21 | tzanger | thirdparty what? |
00:31.39 | doughecka_ | you downloaded the thirdparty version of firefly correct? |
00:31.42 | tzanger | yes |
00:31.55 | tzanger | 1.9.5 build 3935 |
00:32.11 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
00:32.14 | doughecka_ | huh |
00:32.17 | tzanger | in the about page it shows a bunch of 3rdparty links too |
00:32.22 | PoWeRKiLL | anyone using zaptelrtc ? |
00:32.23 | ManxPower | doughecka: If you continue to have problems you can /msg me. You've helped enough people for me to spend time to help you. Next time don't install a server the day before it goes into production |
00:32.24 | crdobbs | I am an * noob. Is there a codec that doesnot bucher FAX transmissions? |
00:32.30 | tzanger | uses libiax2... uses vivida licence... uses speex... |
00:32.38 | ManxPower | crdobbs: ULAW and ALAW are the only ones |
00:33.03 | crdobbs | dont thoes suck up a lot of bandwidth |
00:33.04 | PoWeRKiLL | crdobbs: and you have to get a perfect bandwith :) |
00:33.08 | Phantom-X | damn asterisk seems really cool =) |
00:33.24 | tzanger | doughecka_: I have route my internal and external calls using this netowrk checked, is that correct? |
00:33.44 | tzanger | if I uncheck all codecs I get an error when I try to call saying it can't negotiate codec |
00:33.52 | doughecka_ | yes |
00:34.00 | tzanger | NOTICE[131080]: chan_iax2.c:5664 socket_read: Rejected connect attempt from 192.168.1.105, requested/capability 0x0/0x0 incompatible with our capability 0x2. |
00:34.04 | tzanger | and that in the * log which is correct |
00:34.07 | doughecka_ | [2006] |
00:34.11 | doughecka_ | type=friend |
00:34.15 | doughecka_ | secret=blah |
00:34.18 | doughecka_ | host=dynamic |
00:34.22 | doughecka_ | notransfer=yes |
00:34.23 | crdobbs | what codec has the best quality vs bandwidth ratio |
00:34.25 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
00:34.26 | doughecka_ | thats what I am using |
00:34.31 | doughecka_ | for my iax.conf for my firefly client |
00:34.32 | doughecka_ | and it works |
00:34.35 | tzanger | crdobbs: I like gsm for free and g729 for pay |
00:34.39 | tzanger | doughecka_: yeah that's what I have too |
00:34.41 | redder86 | JunK-Y: I'm thinking about parsing the 'asterisk -r -x "show channels"' output in my AGI script that I already use for caller*id-based call routing |
00:35.00 | Docelm0 | Anyone in here done any AGI PHP scripting? |
00:35.02 | redder86 | JunK-Y: so if the extension is in-use, then it will know |
00:35.02 | crdobbs | what aboud ADPSM, has anyone used that? |
00:35.20 | crdobbs | excuse me ADPCM |
00:35.24 | bkw_ | evil |
00:35.25 | bkw_ | use g726 |
00:35.30 | bkw_ | ADPCM sounds like ass |
00:35.51 | crdobbs | that good, ha :) |
00:35.52 | tzanger | bkw_: no, lpc10 sounds like ass |
00:36.07 | tzanger | doughecka_: I have verified that it is only letting gsm through which is right |
00:36.14 | bkw_ | speex 2,3 sounds EXCELLENT |
00:36.26 | tzanger | (I tried with the other checked one at a time and it failed to negotiate whcih is correct) |
00:36.30 | tzanger | but yeah |
00:36.32 | tzanger | no audio |
00:36.34 | tzanger | grrr |
00:36.41 | tzanger | xlite works fine but I hate xlite |
00:36.50 | crdobbs | thankyou for the info. |
00:36.58 | doughecka_ | tzanger: huh... :( |
00:36.59 | tzanger | haven't played with speex |
00:37.06 | tzanger | what's the 2,3 behind that, bkw_ ? |
00:38.53 | ariel_ | Has any one worked with a lucent Vina-intergrator Channel Bank here? |
00:39.32 | file[laptop] | Paypal is quite nice on 'da phone... |
00:39.50 | file[laptop] | "Can you check to see if my replacement card is being mailed out?" "Can you remove these preauth charges?" |
00:40.06 | tzanger | doughecka_: I get no audio at all (no ringback) if I tell firefly to use SIP |
00:40.13 | tzanger | oh wait |
00:40.26 | Docelm0 | Can anyone help me? Im looking for a little help with the whole PHP AGI script deal.. |
00:40.29 | tzanger | I get no ringback but I get audio once hte call is established |
00:40.30 | doughecka_ | hmm |
00:40.40 | tzanger | IAX2: ringback but no call audio |
00:40.47 | tzanger | SIP: no ringback but call audio (and it's crappy quality) |
00:40.54 | tzanger | no other changes |
00:42.58 | doughecka_ | HAHA |
00:42.59 | doughecka_ | wierd |
00:43.14 | habakuk | docelm0 what are you trying to do? |
00:43.36 | doughecka_ | tzanger: I would check my own config in firefly but I am not booted into windows |
00:44.11 | *** join/#asterisk RageMax (~max@dhcp-064-247-097-106.eg4.ohiou.edu) |
00:44.28 | PoWeRKiLL | removing rtc from my kernel completely broke up playbacking audio file from * is that normal ? |
00:44.43 | RageMax | is that linux ip phone out yet that supports IAX out of the box? |
00:44.47 | Docelm0 | Get the variables etc.. Just screw around with the interface to see what I can do with it. |
00:44.57 | tzanger | doughecka_: no it's not that |
00:45.01 | Docelm0 | Like how do I pass var's to php and pass them back etc.. |
00:45.01 | tzanger | I'm sure of it |
00:45.05 | ManxPower | PoWeRKiLL: for cvs -head anythng is possible, for cvs -stable that should not happen |
00:45.17 | tzanger | I did a make update I saw some interesting changes over the last few days |
00:45.47 | PoWeRKiLL | ManxPower I'm currently using -head |
00:45.48 | doughecka_ | hmm |
00:46.00 | ManxPower | PoWeRKiLL: Do you need the -head features? |
00:46.23 | habakuk | PoWeRKiLL, huh I had the same problem recently. why Do you think it is caused by rtc? |
00:46.24 | PoWeRKiLL | I always use -head so I don't know what the stable don't have :( |
00:47.02 | ManxPower | PoWeRKiLL: Don't use -head then. |
00:47.15 | ManxPower | As of 1.0 -stable is what *I* think people should use for production |
00:47.44 | *** join/#asterisk Inv_arp (junya@adsl-10-164-152.mia.bellsouth.net) |
00:47.50 | PoWeRKiLL | habakuk cause I wanted meetme in my * box, but I can't use ztdummy I have usb-ohci, so I remove rtc from my kernel |
00:48.47 | habakuk | PoWeRKiLL, right I have the same issue. so you are using zaprtc? and seeing this? when was the last time you updated your code? |
00:48.47 | PoWeRKiLL | and compile zaptelrtc I load it but I don't have any audio on MOH Voicemail or other playback |
00:49.20 | PoWeRKiLL | habakuk |
00:49.21 | PoWeRKiLL | CVS-HEAD-09/20/04-23:17:39 |
00:49.26 | PoWeRKiLL | and you ? |
00:49.42 | habakuk | PoWeRKiLL, right I had the same problem, but when I disabled conferencing it all started working again.. didn't look at it too seriously though |
00:49.54 | habakuk | mine is from august sometime |
00:50.30 | PoWeRKiLL | what do you mean you disable conferencing you are just not dialing to meetme app or you remove the apps ? |
00:51.40 | habakuk | PoWeRKiLL, hmm.. I think all of the above I noticed that it didn't show up until I called the conference. So I just disabled the app, so I didn't call by accident |
00:52.10 | tzanger | damn I get shit audio quality out of firefly, period |
00:53.05 | PoWeRKiLL | interesting I'm currently download 1-0 from cvs to see if the problem get resolve |
00:53.27 | PoWeRKiLL | if not I will try to remove the app and if not I will get back RTC in my kernel |
00:54.22 | tzanger | ahh there better quality |
00:54.25 | tzanger | but still weirdness heh |
00:54.26 | ariel_ | I have a problem with a channel bank. I don't get a ring. I can take calls if I know there calling, I can make calls without problems ARgh. |
00:54.41 | tzanger | ariel_: what channel bank |
00:54.44 | ariel_ | Anyone know if signally should be loop start or ground. |
00:54.53 | ariel_ | Lucent Vina |
00:55.53 | ariel_ | Why can't they all just work out of the box like the Adtrans. |
00:58.00 | ManxPower | ariel_: I think loopstart is the default for most things |
00:58.04 | PoWeRKiLL | so habakuk currently you have or you don't have rtc support in your kernel ? |
00:58.28 | habakuk | PoWeRKiLL, no rtc support at the moment |
00:59.26 | doughecka_ | tzanger: works? |
00:59.30 | PoWeRKiLL | at all ? |
01:02.44 | tzanger | doughecka_: better audio quality but still same problem as before |
01:02.56 | doughecka_ | huh. |
01:03.14 | habakuk | PoWeRKiLL, zaprtc is loaded |
01:03.24 | doughecka_ | if it works on a cold boot |
01:03.29 | doughecka_ | I will eat my hat |
01:04.06 | doughecka_ | dang it |
01:04.06 | ariel_ | ManxPower, that is what set it to. But it does not ring.. Argh. I need to get a shoot gun and shoot it. |
01:04.07 | doughecka_ | it does work |
01:04.16 | Jason357 | what is the smallest linux distro astrisk esily installs on? Does it install well on DamnSmallLinux.org? |
01:04.18 | doughecka_ | ariel_: shot gun? :) |
01:04.39 | tzanger | Jason357: I have it running off slack91 in under 500M and that wasn't even trying to shrink it |
01:04.48 | ariel_ | that way they get something like an Adtran or a CAC which I know work. |
01:07.33 | PoWeRKiLL | right now no |
01:11.36 | PoWeRKiLL | ManxPower I try -stable and still same problem |
01:12.55 | ManxPower | PoWeRKiLL: What does "show version" show? |
01:13.09 | Connor_ | ~seen kram |
01:13.13 | jbot | kram is currently on #asterisk. Has said a total of 4 messages. Is idling for 3h 42m 7s |
01:13.20 | PoWeRKiLL | ManxPower: Asterisk CVS-v1-0-10/08/04-02:54:28 built by root@sip on a i686 running Linux |
01:15.50 | PoWeRKiLL | I think I will recompile back the kernel with RTC |
01:15.54 | *** join/#asterisk Poemius (~poemius@adsl-70-48-192-81.adsl.iam.net.ma) |
01:16.27 | Poemius | hi everyone |
01:17.48 | *** join/#asterisk AsteriskJacob (~trillian@md-frederick-bw1-68-170-244-136.chvlva.adelphia.net) |
01:18.22 | habakuk | PoWeRKiLL, hmm are you running a 2.6 / 2.4 kernel? |
01:18.42 | PoWeRKiLL | habakuk 2.4 |
01:19.04 | AsteriskJacob | hey peeps! I like to use the Cvsup script from the asterisk.org download page... Does anyone no how to modify it to get the 1.0 branch? |
01:19.21 | habakuk | PoWeRKiLL, I heard that it's supposed to work very well with 2.6.. but I haven't tried it yet |
01:19.21 | AsteriskJacob | I thought *default release=cvs tag=1.0 would work... |
01:19.27 | AsteriskJacob | but it doesn't seem to |
01:20.00 | habakuk | PoWeRKiLL, zaprtc that is |
01:21.07 | *** join/#asterisk MicroChip32 (~mc@DSL-49-70.che.centurytel.net) |
01:21.08 | Poemius | anyone successfully use distinctive ring on incoming sip calls? |
01:21.24 | Poemius | (to route call to a different extension) |
01:21.57 | outtolunc | AsteriskJacob: why not mirror the whole thing? |
01:21.59 | PoWeRKiLL | habakuk may be but I don't think that I will upgrade a major kernel version now on a production server :) |
01:24.06 | MicroChip32 | is it possible to make sip work, reliably, behind nat. here's the situation: i use wifi at assorted hotspots, which are generally natted. I would like to be able to use my sip device (pap2) at these locations. |
01:27.14 | PatrickDK | microchip, only if they support stun, and you support stun, will that work |
01:28.17 | PoWeRKiLL | my kernel is still compiling hope it will work i put the native RTC as module so I can load it and unload it easyly |
01:28.20 | bkw_ | AsteriskJacob its tag=v1-0 |
01:29.05 | MicroChip32 | PatrickDK: 'they' refers to the hotspot ?? |
01:29.15 | *** join/#asterisk jeb-c4 (~jeb-c4@pcp03913576pcs.nash01.tn.comcast.net) |
01:29.20 | AsteriskJacob | just noticed that... thanks |
01:29.27 | PatrickDK | yep |
01:30.34 | MicroChip32 | and me supporting stun refers to my sip device (which does) or the voip server (which i dunno, its asterisk) |
01:30.40 | jeb-c4 | just curious as to what CVS revision to track for stable? (It appears that v1-0_stable is dead) |
01:30.59 | PatrickDK | na, I think just the nat box and your box have to support stun |
01:31.37 | MicroChip32 | "your box" = server (asterisk) right ? |
01:31.53 | PatrickDK | your box = whatever your using on the hotspot |
01:32.12 | bkw_ | Vonage is planning to use MP3 playback technology for voice mail soon, according to company CEO |
01:32.16 | MicroChip32 | ah ok, i know my sip device does. who knows about the hotspot. the one i know is a linksys, no idea about the others |
01:32.24 | bkw_ | OK I WANT MP3 playback of voicemail by tommorow |
01:32.55 | icebalm | can't do it, patent encumbered, we can do ogg vorbis tho |
01:32.56 | PatrickDK | mp3 playback? |
01:32.58 | PoWeRKiLL | bkw_ Vonage use * for their voicemail right ? |
01:33.09 | PatrickDK | why would that be a customer benifit? |
01:33.14 | hmodes | aww comon bkw, give us our moment ;p |
01:33.17 | hmodes | it took long enough |
01:33.22 | PatrickDK | I only see that as a benifit to the company |
01:33.45 | outtolunc | you should already be able to do mp3->ice->* <G> (ducking) |
01:33.48 | hmodes | (hi, by the way) |
01:34.12 | amer | Vonage use * for their voicemail right ? is this true? |
01:35.02 | bkw_ | hmodes we will have it by morning |
01:35.05 | bkw_ | so na na ne boo boo |
01:35.17 | bkw_ | accually anthm and I are very pissed off at the voicemail in asterisk |
01:35.23 | hmodes | damn! |
01:35.25 | bkw_ | expect some new shit to come from the anthm factory |
01:35.41 | hmodes | amer: http://investorrelations.utstar.com/ReleaseDetail.cfm?ReleaseID=137677 |
01:36.46 | Phantom-X | do you guys know why emerge asterisk would want to install apache ( I have strong aversiona gainst webservers ) ? |
01:38.22 | amer | hmodes: so this means they are not using * for voicemail |
01:38.25 | bkw_ | emerge -pv asterisk |
01:38.42 | bkw_ | Phantom-X do not EMERGE asterisk |
01:38.45 | bkw_ | you do it froms rc |
01:38.50 | bkw_ | the portage is so out of whack its not even funny |
01:39.05 | bkw_ | its 0.9.0 |
01:39.08 | bkw_ | its OLD OLD OLD |
01:39.20 | amer | can I apply patch for the latest cvs build to a month old build? |
01:39.21 | Phantom-X | hmm |
01:39.24 | bkw_ | Phantom-X do you not know how to use USE flags? |
01:39.29 | Phantom-X | the put up something newwer |
01:39.43 | PoWeRKiLL | habakuk it's working :) The problem was the RTC no meet me but everything works back |
01:39.47 | Phantom-X | bkw_, yes and still even if I have -apache and -apache2 |
01:39.54 | icebalm | bkw_: 1.0 is in portage |
01:39.57 | Phantom-X | it tries to emerge that mongoloid apcahe |
01:40.17 | Phantom-X | I really hate to have a webserver on my host |
01:40.20 | bkw_ | emerge -pv |
01:40.29 | bkw_ | emerge -pv asterisk |
01:40.31 | Phantom-X | you trie emerge -upv |
01:40.35 | bkw_ | no u |
01:40.36 | bkw_ | u bad |
01:40.50 | icebalm | -u is not bad |
01:41.02 | bkw_ | it sure can be |
01:41.09 | Phantom-X | and saw im supposed to be able to do USE="-apache2" but it still persist in instaling apache |
01:41.09 | icebalm | only if you're a bloody retard |
01:41.15 | bkw_ | haha no |
01:41.21 | icebalm | bkw_: how can it be bad |
01:41.25 | bkw_ | export USE="-apache2" |
01:41.42 | Phantom-X | [ebuild N ] net-misc/asterisk-1.0.0 +alsa -apache2 -doc +gtk -mmx -mysql -nopri -nozaptel 10,993 kB |
01:41.44 | bkw_ | icebalm I haven't run into the problem with u but someone else in here informed me |
01:41.52 | bkw_ | Phantom-X its not going to install apache |
01:41.52 | Phantom-X | im not haing to export it at all |
01:42.12 | bkw_ | Phantom-X you export your USE flags |
01:42.16 | Phantom-X | you should just do : USE="-apache2" emerge asterisk |
01:42.21 | icebalm | bkw_: it's not bad, there's no senario where it can harm anything, unless you're not using portage properly |
01:42.28 | bkw_ | maybe it was U |
01:42.43 | Phantom-X | well |
01:42.45 | PoWeRKiLL | ManxPower habakuk I found the problem you have to compile kernel RTC as module not as NO |
01:42.45 | Phantom-X | I checked : |
01:42.57 | bkw_ | Phantom-X never install asterisk from portage |
01:42.59 | Phantom-X | USE="-apache2" emerge -upv asterisk |
01:43.02 | bkw_ | never install mpg123 from portage |
01:43.04 | Phantom-X | still wanted to install apache |
01:43.08 | bkw_ | you're doing it wrong |
01:43.14 | Phantom-X | no im not |
01:43.14 | bkw_ | export USE="-apahce2" |
01:43.16 | bkw_ | then emerge |
01:43.23 | Phantom-X | thats the way you are suppose to do with emerge |
01:43.43 | bkw_ | you can export it to the USE var in your env too you ninny |
01:43.46 | bkw_ | don't listen to me then |
01:43.51 | bkw_ | I do this all the time |
01:43.55 | icebalm | Phantom-X: asterisk doesn't depend on apache, it depends on mpg123 and newt, the rest are configurable via USE |
01:44.03 | Phantom-X | still trying to install apache |
01:44.09 | Phantom-X | I did your variant also |
01:44.14 | *** part/#asterisk AsteriskJacob (~trillian@md-frederick-bw1-68-170-244-136.chvlva.adelphia.net) |
01:44.23 | Phantom-X | still emerge wants to install apache |
01:44.26 | bkw_ | if it has a -apache2 when you do emerge -pv |
01:44.30 | bkw_ | then its not going to do apache |
01:44.59 | icebalm | Phantom-X: when you do emerge -vp asterisk what packages does it try to install? |
01:45.35 | icebalm | the -apache2 use flag is moot, the asterisk ebuild doesn't even look at apache |
01:45.51 | bkw_ | <Phantom-X> [ebuild N ] net-misc/asterisk-1.0.0 +alsa -apache2 -doc +gtk -mmx -mysql -nopri -nozaptel 10,993 kB |
01:45.51 | Phantom-X | icebalm, I tries to instal some packages |
01:45.56 | Phantom-X | and one of them is apache |
01:46.06 | bkw_ | then something else that you need is depending on apache |
01:46.09 | bkw_ | along the way |
01:46.13 | icebalm | that does it install right after apache? |
01:46.15 | bkw_ | but that shouldn't |
01:46.21 | icebalm | what does it install right after apache? |
01:46.35 | Phantom-X | bkw_, well the list tells non is but asterisk |
01:46.45 | bkw_ | the -apache2 doesn't mean shit |
01:46.48 | Phantom-X | but I put -apache2 so its not supposed to instal |
01:46.56 | bkw_ | if you see -apache2 then you're good to go |
01:47.04 | bkw_ | show me the full output of emerge -pv asterisk |
01:47.08 | icebalm | you'll also want -apache |
01:47.16 | bkw_ | put it on pastebin.ca |
01:47.27 | bkw_ | I personally do not use portage for anythign asterisk related |
01:47.40 | bkw_ | mpg123 from portage is fucked |
01:48.15 | icebalm | uh, howso? |
01:48.20 | Phantom-X | http://pastebin.ca/1324 |
01:48.37 | Phantom-X | ive never had any probs with portage |
01:48.55 | Phantom-X | until now with asterisk , im certain that the ebouild is wrong built |
01:49.01 | bkw_ | your portage is smokin crack |
01:49.14 | bkw_ | icebalm anything but 0.59r is broken |
01:49.20 | Phantom-X | bkw_, blame the person who made the ebould for astersik |
01:49.20 | bkw_ | it gives strange results |
01:49.23 | Phantom-X | I could check |
01:49.37 | bkw_ | my emerge sync from today is doing 0.9.0 |
01:49.40 | bkw_ | odd |
01:49.46 | bkw_ | the 1.0 ebuild is fucked up then |
01:49.54 | Phantom-X | bkw_, change sync server |
01:49.55 | brc_ | suprise! |
01:49.58 | brc_ | it's gentoo |
01:50.10 | bkw_ | gentoo on a server ROCKS |
01:50.10 | Damin | Discovery has a 1 hour show on SpaceShip One. |
01:50.12 | brc_ | use debian |
01:50.15 | Damin | Gentoo is lame. |
01:50.22 | Damin | Gentoo should be banned. |
01:50.23 | brc_ | in debian we had 1.0 THREE MONTHS AGO! |
01:50.25 | brc_ | beat that! |
01:50.30 | icebalm | bkw_: it's not fucked up, it's masked, geez, you have a great tendancy to say things are fucked up or broken when they really arent |
01:50.40 | brc_ | Damin, when is that on? |
01:50.41 | brc_ | now? |
01:50.42 | bkw_ | icebalm fuck you |
01:50.51 | Damin | icebalm: He is a drama queen. :) |
01:50.52 | brc_ | calm down bkw_ |
01:50.55 | Damin | Yeah.. it's on now.. |
01:50.57 | bkw_ | brc_ fuck you |
01:51.02 | Damin | Last ten minutes.. |
01:51.05 | icebalm | Damin: so it would seem |
01:51.33 | Damin | It's called "Black Sky" |
01:51.37 | bkw_ | icebalm no I just don't pay much attention when its not my problem |
01:51.40 | bkw_ | and thats the truth |
01:51.40 | Damin | Lots of awesome footage.. |
01:52.02 | icebalm | bkw_: fair enough, but you shouldn't say things are "fucked up" when they are working as intended, it's FUD |
01:52.14 | bkw_ | 0.59s is fucked |
01:52.16 | bkw_ | i know that for sure |
01:52.34 | *** join/#asterisk denon (denon@synapse.subneural.net) |
01:52.34 | *** mode/#asterisk [+o denon] by ChanServ |
01:53.11 | brc_ | thanks Damin ...Scientific Atlanta'ing it |
01:53.30 | brc_ | oh...it's on at 7pm here |
01:53.38 | icebalm | Phantom-X: the only thing I can think of is that the asterisk ebuild does make mention of a voicemail webapp, but it doesn't (or shouldnt) get installed if you dont have a webserver |
01:54.06 | bkw_ | icebalm that was my first thought |
01:54.18 | bkw_ | but I dont have the ebuild yet |
01:54.19 | bkw_ | so I can't see |
01:54.34 | brc_ | the voicemail 'webapp' is called 'webvmail' |
01:54.44 | brc_ | if you have the asterisk source you can 'make webvmail' iirc |
01:54.50 | Phantom-X | icebalm, right , and I have no webserver or ever will install one such since I hate them =) |
01:55.31 | bkw_ | Phantom-X this is why I don't use ebuilds for asterisk |
01:55.31 | bkw_ | they have been so messed up since day one |
01:55.32 | file[laptop] | or... day -1! |
01:55.39 | Damin | What is so hard about typing "make" |
01:55.46 | Phantom-X | bkw_, im not listening to that |
01:56.06 | bkw_ | Phantom-X good luck |
01:56.15 | bkw_ | trust me compile it yourself |
01:56.18 | Phantom-X | bkw_, I will make it =) |
01:56.19 | bkw_ | if youw ant packages use Debian |
01:56.35 | Phantom-X | debian is crap , using your own words =) |
01:56.39 | bkw_ | yep |
01:56.40 | Damin | If you want problems, use Gentoo! :) |
01:56.43 | bkw_ | I totally agree |
01:56.47 | bkw_ | doh |
01:56.48 | bkw_ | you ass |
01:56.50 | Phantom-X | gentoo is fine |
01:56.52 | bkw_ | No gentoo isn't a problem |
01:56.58 | Phantom-X | been using if for many years |
01:57.02 | bkw_ | now whats fun is when it fucks up your gcc-config during emerge |
01:57.14 | bkw_ | now thats a fun one when it happens to ya for the first time |
01:57.30 | Phantom-X | bkw_, are you rally making sense ? |
01:57.51 | Phantom-X | bkw_, why use a computer in the first pace then ? |
01:58.42 | bkw_ | um |
01:58.47 | bkw_ | wtf did I do to you? |
01:58.48 | mishehu | bah, hylafax is pissing me off. |
01:58.53 | bkw_ | hylafax is easy |
01:58.56 | file[laptop] | 9..9..6! |
01:59.00 | Damin | I prefer using an Abacus myself.. |
01:59.20 | *** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net) |
01:59.34 | file[laptop] | paulc: 9..9..6! |
02:00.06 | mishehu | bkw_: I constantly get "Waiting for modem to come free", my modem is on ttyS1, and faxgetty launches on ttyS1... |
02:00.36 | mishehu | nothing ever changes, and I've tried multiple configurations. the linksys pap2 has nothing to do with it. |
02:01.06 | icebalm | nah, what's fun is when you apt-get dist-upgrade and then you go to login and your keyboard isn't inputting anything because some idiot debian package manager compiled /bin/login without local console support |
02:01.49 | *** join/#asterisk Mike (~mike@201.135.48.52) |
02:02.25 | mishehu | in fact, I got the pap2 to be able to test out hylafax. |
02:02.29 | *** join/#asterisk intrin (~fasdfa@c66.188.173.60.stc.mn.charter.com) |
02:02.47 | intrin | paulc! |
02:02.48 | intrin | :D |
02:03.01 | paulc | evening all.. |
02:03.04 | *** join/#asterisk PhilM (~a@r42h98.res.gatech.edu) |
02:03.09 | intrin | you just get here too? |
02:03.21 | paulc | yeah, few mins ago |
02:03.24 | intrin | cool |
02:03.26 | intrin | perfect timing |
02:03.40 | intrin | man, watched cabin fever |
02:03.43 | intrin | boy was that gay |
02:04.02 | bkw_ | haha |
02:04.11 | mishehu | bkw_: you able to help at all? |
02:04.13 | JerJer | Power over Wireless Ethernet |
02:04.19 | bkw_ | mishehu does the modem work from minicom? |
02:04.25 | mishehu | bkw_: *nod* |
02:04.37 | mishehu | it's a USR ISA modem |
02:04.41 | *** join/#asterisk invi_ (~undisclos@dsl-cap-209-5-169-185-cgy.nucleus.com) |
02:04.46 | bkw_ | evil |
02:04.47 | file[laptop] | JerJer: power over bagel over wifi over toast over dingbats |
02:04.56 | bkw_ | mishehu not sure never used an ISA modem wif it |
02:05.14 | mishehu | bkw_: I thought winmodems were evil, not per-say ISA hardware modems. ;-) |
02:05.16 | invi_ | is this the only way of doing Call-Waiting flash > http://lists.digium.com/pipermail/asterisk-users/2004-September/060960.html |
02:05.30 | bkw_ | mishehu I used a 48 port Patton 2977 card |
02:05.47 | mishehu | bkw_: appearantly not |
02:05.48 | bkw_ | 48 modems on that bad boy |
02:05.49 | mishehu | heh |
02:06.00 | mishehu | little warfaxing going on there? |
02:06.00 | mishehu | hehe |
02:06.05 | bkw_ | just a bit |
02:06.08 | bkw_ | :) |
02:06.12 | *** join/#asterisk Stealth_Man (~no@ool-18bc203f.dyn.optonline.net) |
02:06.13 | bkw_ | FAXBLAST |
02:06.26 | bkw_ | I got tired of waiting on app_rxfax/txfax |
02:06.35 | bkw_ | and 9600bps is for the birds |
02:06.39 | mishehu | well, i'd put a PCI USR hardware modem on there, but I have no free PCI slot |
02:06.45 | bkw_ | I have one of those too |
02:06.47 | bkw_ | that works great |
02:06.52 | ManxPower | Alcohol and Calculus don't mix. Don't drink and derive. |
02:07.00 | bkw_ | OLD |
02:07.17 | mishehu | and my X101P is sharing interrupts with the usb controller already |
02:07.45 | mishehu | oh and I don't have a serial cable for my external USR modem |
02:07.46 | florz | What's the state of app_*fax |
02:07.48 | florz | BTW? |
02:07.50 | florz | gnah |
02:07.54 | bkw_ | florz CRAP 1.0 |
02:07.55 | mishehu | gnaw |
02:09.01 | florz | bkw_: Hmm ... "it might work under unknown circumstances"? |
02:09.22 | ManxPower | florz: .01 had problems sending to Canon fax machines. |
02:09.53 | mishehu | I wonder when the SHIT 2.0 version will be out. |
02:10.01 | florz | ManxPower: That sounds more like "works most of the time but is not stable"!? |
02:10.03 | bkw_ | dont know but I don't have tow orry with it |
02:10.13 | bkw_ | app_txfax_and_segfault.c |
02:10.43 | jgaviria | im installing zapata in debian with kernel 2.4.26 but i have unresolved symbols when i tried to load zapata module... somebody have an idea.. somebody could helpme? |
02:10.50 | florz | how about receiving faxes? |
02:11.07 | *** join/#asterisk Wi_Fi (~OUT@c-66-229-160-14.we.client2.attbi.com) |
02:11.35 | jgaviria | i would say zaptel |
02:11.42 | ManxPower | .02 seems to do better |
02:11.47 | Wi_Fi | heyaz |
02:12.10 | intrin | ~google sip.conf.sample |
02:12.13 | Stealth_Man | ~showtime |
02:12.35 | Stealth_Man | ~seen showtime |
02:12.36 | jbot | showtime <show@65.171.196.4> was last seen on IRC in channel #asterisk, 5d 1h 27m 43s ago, saying: 'dumb proxy and registrar'. |
02:12.42 | florz | ManxPower: txfax -> Canon, you mean? |
02:12.56 | ManxPower | florz: Yes. |
02:12.58 | bkw_ | BIG IS BAD |
02:13.01 | Wi_Fi | ~google iax.conf.sample |
02:13.09 | bkw_ | Does size matter? |
02:13.14 | Stealth_Man | ~google asterisk forums |
02:13.18 | bkw_ | and how far would you go to change the lenght or size of your goodies? |
02:13.21 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
02:13.46 | florz | Well, actually receiving faxes would be more important for now, as it could save some paper =:-) |
02:14.43 | Phantom-X | ok I managed to make sense with the emerge =) |
02:15.09 | Wi_Fi | i need help gettin iaxy in dialplan |
02:15.15 | *** join/#asterisk Moc (~mochouina@modemcable021.49-80-70.mc.videotron.ca) |
02:15.20 | bkw_ | Dial(IAX/iaxy) |
02:15.22 | bkw_ | NEXT!!! |
02:15.45 | *** join/#asterisk lattera (~lattera@dialup-4.228.195.82.Dial1.Denver1.Level3.net) |
02:16.00 | Wi_Fi | make it default ring from zap incoming |
02:16.17 | lattera | is there a list of asterisk-compatible VoIP clients anywhere? |
02:17.29 | Wi_Fi | lattera try firefly |
02:17.35 | Wi_Fi | very good with iax |
02:17.41 | tzanger | Wi_Fi: I can't get it to work |
02:17.44 | lattera | and, will VoIP work well with dialup? |
02:17.55 | lattera | (I'm quite the newb when it comes to VoIP) |
02:18.02 | Wi_Fi | tzanger ??? |
02:18.25 | *** join/#asterisk xpasha (~pavel@217.30.252.68) |
02:18.30 | tzanger | Wi_Fi: with identical settings, I get ringback with IAX but no audio once connected... and changeing from iax to sip I get no ringback but audio |
02:18.50 | tzanger | I use gsm codec (and can verify that if I change codecs, * complains, as I only allow gsm with iax and sip) |
02:18.53 | mishehu | haha that's whacked |
02:18.56 | tzanger | xlite works just fine |
02:19.19 | xpasha | anybody alive who worked with E1/T1 cards? |
02:19.22 | tzanger | mishehu: tell me about it :-) |
02:19.24 | tzanger | xpasha: yup |
02:19.25 | tzanger | T1 |
02:19.28 | *** join/#asterisk afrosheen (~afro@c-67-166-162-197.client.comcast.net) |
02:19.29 | tzanger | T100P and TE405P |
02:20.06 | xpasha | tzanger so may be you know what is to be done in order to make E1 card being a clock source for remote device? |
02:20.10 | *** join/#asterisk Kb1_Kanob (johnsmith@sec2d47.dial.uniserve.ca) |
02:20.17 | tzanger | xpasha: it won't |
02:20.20 | tzanger | well it can |
02:20.29 | xpasha | how? |
02:20.32 | tzanger | but it'll clock off the internal timer which I wouldn't trust with a 10 foot pole |
02:20.46 | tzanger | well in any link you have a clock source |
02:21.02 | xpasha | I have cisco 5350 and it can't generate clock signal |
02:21.02 | *** join/#asterisk voiper (~none@pcp09528872pcs.eatntn01.nj.comcast.net) |
02:21.07 | tzanger | if the card is not syncing to the other side, then it's running free |
02:21.20 | tzanger | yeah the 5350 won't clock you're right |
02:21.33 | xpasha | so how to force it to generate signal? |
02:21.34 | tzanger | so tell the digium card not to sync from the line |
02:21.41 | tzanger | you don't you jus ttell it not to sync ot hte line |
02:21.42 | xpasha | what option? |
02:21.44 | *** join/#asterisk telme (~non@c-24-8-57-124.client.comcast.net) |
02:21.47 | tzanger | span= |
02:21.58 | tzanger | read the zaptel.conf it's listed in there |
02:21.59 | xpasha | ? |
02:22.04 | tzanger | you just say 0 = don't use the line for clock sync |
02:22.12 | Kb1_Kanob | span=1,0,0 to push timing, 1,1,0 to pull timing. |
02:22.18 | tzanger | um |
02:22.18 | xpasha | hmmm |
02:22.18 | tzanger | no |
02:22.22 | tzanger | Kb1_Kanob: that is not true |
02:22.26 | xpasha | so |
02:22.30 | xpasha | listen to me guys |
02:22.42 | xpasha | I have done span=1,0,0 and span=1,1,0 |
02:22.53 | xpasha | and anyway |
02:23.17 | xpasha | It wroked well with remote PBX I connected |
02:23.18 | Kb1_Kanob | 0 for push and 1 for pull, according to digium support on t100p. |
02:23.19 | tzanger | the card has an internal clock; you can have it try to sync to incoming line or not. span=1,0,0 says not to try and sync, 1,1,0 says to try and use recovered clock as primary sync src |
02:23.44 | voiper | does any know about this message ?¿ "ast_rtp_read: Unknown RTP codec 72 received" |
02:23.45 | xpasha | I have PRI trunk from Alcatel S12 PBX |
02:23.48 | tzanger | there really is no "pushing" of the clock signal, only an attempt to lock ot the recoverd clock |
02:23.50 | xpasha | and it generates clock |
02:23.55 | tzanger | xpasha: is this a 4-span card? |
02:23.57 | xpasha | and anyway |
02:24.04 | xpasha | it worked well |
02:24.09 | xpasha | with my E100P |
02:24.14 | Kb1_Kanob | tzanger: I take it you're quite familiar w/internals - can you answer some bus questions? |
02:24.18 | tzanger | xpasha: everything generates clock, don't think of it that way |
02:24.26 | tzanger | Kb1_Kanob: might be able to :-) |
02:24.26 | xpasha | if I set span=1,0,0 it worked |
02:24.31 | xpasha | if I set span=1,1,0 it worked too |
02:24.35 | tzanger | xpasha: stop |
02:24.38 | xpasha | who to explain it? |
02:24.38 | lattera | are there any test PBXs available? |
02:24.38 | tzanger | xpasha: listen to me |
02:24.52 | tzanger | xpasha: all T1/E1 devices self-clock.. ALL of them |
02:24.56 | tzanger | the 5350 will |
02:25.20 | tzanger | xpasha: the problem is that if you have two sides both clocking to different sources (i.e. themselves) you will get frame slips |
02:25.32 | florz | lattera: Depending on what you want to test, I'd say: Just install Asterisk. |
02:25.33 | Kb1_Kanob | tzanger: relative clock drift. |
02:25.34 | tzanger | so you designate one side to lock on to the other |
02:25.47 | *** join/#asterisk monzsca (~monzsca@monz.marmoset.net) |
02:25.56 | tzanger | since the 5350 demands that you lock on to the other side, you tell * not to try and lock on to the other side |
02:26.03 | tzanger | and therefore you use a span line of 1,0 |
02:26.04 | voiper | does anyone know about this message "ast_rtp_read: Unknown RTP codec 72 received" |
02:26.16 | tzanger | (span 1, don't try to lock on to recovered clock) |
02:26.30 | tzanger | voiper: not me, I don't use RTP unless I can't help it |
02:26.37 | xpasha | tzanger so how to explain that my E100P worked well anyway if I set 1,0,0 or 1,1,0 ?? |
02:26.47 | tzanger | xpasha: because the clocks are pretty much in sync |
02:26.58 | lattera | florz, I would, but I want to test to see how well it would (or would not) work on dialup |
02:27.16 | tzanger | you won't notice desync'd hardware until frame slips happen and if the clocks are worth anything at all that won't happen very often |
02:27.23 | tzanger | but when it odes happen you'll hear a buzz or chirp |
02:27.26 | xpasha | I meant worked with PBX that generates signali itself |
02:27.32 | tzanger | xpasha: yes |
02:27.35 | Wi_Fi | bkw_ would that be IAX2/iaxy? |
02:27.42 | tzanger | xpasha: the one end is going |
02:27.54 | xpasha | tzanger but if I set 1,0,0 it have to generate clock |
02:28.00 | tzanger | __________----------_________----------__________----------__________ |
02:28.02 | tzanger | and the other side is going |
02:28.08 | voiper | thanks tzanger |
02:28.15 | tzanger | __________----------__________----------__________---------__________ |
02:28.17 | xpasha | and I would get no working |
02:28.19 | tzanger | i.e. ALMOST the same |
02:28.28 | tzanger | but you can see there it's starting to slip |
02:28.37 | xpasha | hmmm |
02:28.40 | tzanger | when you get enough of a difference you'll slip a frame |
02:28.47 | tzanger | xpasha: THEY ALL GENERATE CLOCK |
02:28.50 | xpasha | :) |
02:28.51 | tzanger | please understand that |
02:28.57 | tzanger | they ALL generate their own clock signal |
02:29.06 | xpasha | tzanger okok :) |
02:29.15 | tzanger | what you are saying with the span command is wiehterh to try and lock on their own internal clock ot the recovered clock from the line |
02:29.31 | Kb1_Kanob | tzanger: hence 'master'/'slave' terminology rather than 'push'/'pull'? |
02:29.37 | tzanger | the 5350 you have no choice, you have to say whether the incoming line is clock primary or clock secondary (or teritary or quaternary) |
02:29.42 | tzanger | Kb1_Kanob: kind of yeah |
02:29.54 | xpasha | tzanger did you try span=1,0,0 with 5350? |
02:29.58 | tzanger | all you're doing is telling the driver whether to try and use the recovered clock as a sync source |
02:30.02 | xpasha | got it working? |
02:30.13 | tzanger | xpasha: I don't have a 5350, I have a 5248 and an ancient TNT |
02:30.18 | tzanger | er not TNT but MAX |
02:30.21 | tzanger | and it works just fine |
02:30.38 | xpasha | the same situation it can generate clock signal? |
02:30.38 | tzanger | I have a TE405P. span 1 is a PRI from Bell Canada. I say to lock on to that |
02:30.44 | xpasha | it can't I meant |
02:31.01 | tzanger | span 2 I have connected to the Ascend MAX. I tell it NOT to sync there, and I tell the MAX ot sync to the line (which is span 2 on *) |
02:31.13 | xpasha | hmmm |
02:31.14 | xpasha | ok |
02:31.15 | xpasha | :) |
02:31.20 | tzanger | span 3 and 4 go to an Adit600 and again, I tell * NOT to sync to them, but I tell the Adit600 to sync to those lines |
02:31.29 | bkw_ | klasjdfowkdjfoiejf |
02:31.39 | xpasha | I will make cross over and try to get it working with Cisco 5350 |
02:32.01 | xpasha | thanks tzanger :) |
02:32.04 | tzanger | every T1/E1/J1 device generates clock... all you are doing is specifying whether the device should try and lock its clock to the incoming recovered clock |
02:32.25 | Kb1_Kanob | tzanger: so, I understand that the clock on the digium hardware is implemented as a high resoloution hardware timer _but_ if it's the 'master' end then it's succeptible to causing slips when the PCI bus gets contention. However, |
02:32.26 | tzanger | unless there are SERIOUS issues you will always sync up, but if one side is not trying ot match the other you will eventually get frame slips |
02:32.34 | tzanger | no |
02:32.40 | Kb1_Kanob | if it's hardware then it's not the clocking that's slipping, but a buffer underrun? |
02:32.47 | tzanger | the digium timer has nothing to do with the PCI bus |
02:32.54 | xpasha | so what is the option to be a primary clock source, secondary etc? |
02:33.05 | tzanger | the digium timer is on-card and it can either free-run or try and lock to a recovered clock |
02:33.06 | xpasha | to be source for what? |
02:33.10 | Kb1_Kanob | tzanger: I'll let you finish w/xpasha first. sorry. |
02:33.30 | tzanger | xpasha: exactly what it says -- you tell the 5350 how to sync to the incoming T1s |
02:33.49 | xpasha | so what is primary and secondary then? |
02:33.50 | tzanger | you can say controller 1's recovered clock is your primary clock source (or secondary/tertiary/quaternary) and so on |
02:33.56 | *** join/#asterisk infinii (~wayland@host661461427f.dsl.res.tor.fcibroadband.com) |
02:34.03 | tzanger | primary = if I have multiple incoming clocks, use this one first. it's the best |
02:34.13 | tzanger | secondary = if my primary dies goes away, use this one |
02:34.19 | tzanger | tertiary = if my secondary dies, use this one |
02:34.25 | tzanger | quaternary = if my tertiary dies, use this one |
02:34.25 | intrin | ~google x-lite |
02:34.34 | tzanger | just the same as teh span= lines in *... |
02:34.38 | tzanger | 0 = do not sync to this span's clock |
02:34.50 | tzanger | 1 = use this span's clock as the primary clock sync source |
02:34.58 | tzanger | 2 = use this span's clock if the primary clock source is gone |
02:35.08 | tzanger | 3 = use this span's clock if the secondary clock source is gone |
02:35.14 | infinii | good evening |
02:35.19 | tzanger | 4 = use this span's clock if the tertiary clock's source is gone |
02:35.27 | tzanger | pretty straightforward |
02:35.35 | therealadept | "You have entered a number that cannot be reached from your calling area" :/ |
02:35.36 | tzanger | confusing at first, but pretty straightforward after :-) |
02:35.36 | xpasha | Ok understand this :) |
02:35.39 | xpasha | tnx :) |
02:35.40 | tzanger | so |
02:35.55 | tzanger | since the 5350 DEMANDS to sync ot something, yuou tell * NOT to sync to the 5350 |
02:35.56 | xpasha | tzanger what area are you from? |
02:35.59 | infinii | Anyone using the digitnetworks x100p fxo card? seems really affordable |
02:36.00 | tzanger | xpasha: canada |
02:36.06 | xpasha | what province I meant |
02:36.11 | tzanger | ontario |
02:36.14 | xpasha | ah ok |
02:36.20 | tzanger | about an hour and a half from Toronto |
02:36.27 | xpasha | I have a relatives in Quebec |
02:36.48 | xpasha | may be visit canada soon :) |
02:36.58 | tzanger | xpasha: where are you? |
02:37.05 | xpasha | I'm from Russia |
02:37.12 | tzanger | right on |
02:37.18 | tzanger | one of the places I want to visit is Russia |
02:37.19 | xpasha | but work for Czech company |
02:37.33 | xpasha | and visit Prague frequently |
02:37.35 | Wi_Fi | can somebody splain me on adding iaxy to dialplan |
02:38.05 | tzanger | Prague is another place |
02:38.06 | xpasha | tzanger Russia is not so wild as most ppl in the world can imagine :) |
02:38.11 | tzanger | I dunno I like those kinds of places |
02:38.12 | tzanger | :-) |
02:38.21 | tzanger | no I don't want to visit it because it is wild |
02:38.27 | xpasha | hehe $) |
02:38.31 | tzanger | I like the architecture, I like the "old world" I think |
02:38.32 | ariel_ | Wi_Fi, what is your question? |
02:38.42 | Kb1_Kanob | tzanger: may I continue clocking discussion w/you? |
02:38.48 | tzanger | certainly |
02:38.56 | xpasha | tzanger btw Russia is the same nationalities mix as USA and Canada |
02:39.04 | Wi_Fi | ive added it but i cant connect to it from other phones |
02:39.08 | xpasha | I live in the place Tatarstan |
02:39.20 | xpasha | here are ppl tatars |
02:39.20 | tzanger | xpasha: right on |
02:39.22 | tzanger | :-) |
02:39.25 | xpasha | they are muslims |
02:39.26 | therealadept | has anybody been having nufone problems recently? some toll free calls haven't been going through and my callerid is always showing up as unavailable :/ |
02:39.29 | Kb1_Kanob | Ok, so if clock on t100p is hardware then that explains why I get no BPVs on a crossover link, but I do get audio defects during long DMA transfers - because of buffer starvation on the card? |
02:39.30 | tzanger | I want to see Germany, Russia, the Ukraine |
02:39.45 | therealadept | seems like its been happening for the last few days |
02:39.49 | xpasha | my father is ukrainian :) |
02:39.52 | xpasha | btw |
02:39.53 | ariel_ | for real wild things go to Thialand... |
02:39.53 | tzanger | Kb1_Kanob: you likely get audio defects because you're taxing the bus |
02:40.01 | tzanger | are you getting BPV's on DMA transfers? |
02:40.04 | infinii | I'm going to bangcock in Dec |
02:40.09 | tzanger | or is that a red herring |
02:40.10 | infinii | but first japan |
02:40.16 | ariel_ | Wi_Fi, what error are you getting. |
02:40.18 | Kb1_Kanob | no BPVs or alarms detected. |
02:40.31 | tzanger | Kb1_Kanob: ok |
02:40.37 | Kb1_Kanob | but some occasional audio defects when master clock is t100p. |
02:40.48 | Kb1_Kanob | voicemail is running on the host server. |
02:40.52 | ariel_ | infinii, if you want when you get there go to pattia beach it's better. |
02:41.08 | tzanger | your problem is likely bus exhaustion -- try reducing the PCI latency timer for all your PCI devices, and then giving the T100P a higher latency timer value |
02:41.16 | Kb1_Kanob | also getting intermittant timing issues w/RBS signalling on the t1. |
02:41.18 | tzanger | i.e. give it the ability to preempt the other devices on the bus |
02:41.19 | infinii | The extensions.conf that comes with * is kinda full of commented stuff, did you guys simply start from an empty file and build it up to your liking? this std one is hard to read with so much in it |
02:41.31 | Kb1_Kanob | tzanger: setpci? |
02:41.35 | Wi_Fi | ariel_ |
02:41.37 | infinii | ariel_: pattaya is overriden with tourists and isnt as clean as it used to be |
02:41.38 | tzanger | unless you're getting frame slips clocking isn't the problem |
02:41.39 | Wi_Fi | http://www.pastebin.com/108294 |
02:41.44 | tzanger | Kb1_Kanob: yes |
02:42.04 | infinii | ariel_: phuket in thailand has nicer beaches |
02:42.09 | Kb1_Kanob | that makes more sense - your description this evening was extremely good. It belongs in zaptel.conf. |
02:42.31 | tzanger | Kb1_Kanob: :-) |
02:42.36 | tzanger | I just checked my own system |
02:42.43 | tzanger | I have the latency timer set to 64 across the board |
02:42.50 | tzanger | I thought I had the TE405 set a little higher |
02:42.50 | lattera | so, asterisk doesn't run on FreeBSD? |
02:43.03 | tzanger | lattera: it does, I don't think the digium hardware drivers have been ported |
02:43.15 | Kb1_Kanob | On a related note - some services in * consume zaptel clock for timing (eg. signalling). Is it an absoloute timer read or relative? Ie. if there was a bus delay would the signal run on? |
02:43.21 | florz | lattera: *lag* That depends mostly on the codec you are using - which you easiest can test with asterisk's echo application, I think. The other "limiting factor" on dialup (56 k modem?) is latency, which is acceptable ... |
02:43.43 | tzanger | Kb1_Kanob: http://www.reric.net/linux/pci_latency.html |
02:43.53 | Kb1_Kanob | Put another way, could a 200ms wink stretch out to 250ms? |
02:44.07 | florz | lattera: (no, not whether it runs on FreeBSD, but voice quality =:-) |
02:44.14 | Kb1_Kanob | or an incoming 200ms be read as a 150ms? |
02:44.18 | tzanger | Kb1_Kanob: well nothing "consumes" the timer -- it's just a free running timer that can be optionally locked to a recovered clock |
02:44.30 | tzanger | Kb1_Kanob: and what the software uses the timer for is a 1000Hz interrupt |
02:44.47 | *** join/#asterisk enmaca (~emartinez@201.129.118.69) |
02:45.00 | Kb1_Kanob | ahhh.... and it has a tight timing routine tucked away somewhere... |
02:45.16 | tzanger | that is why digium hardware is so picky about having its own IRQ -- if multiple devices have the same IRQ the system must check each driver to see which one the IRQ is really for, and some drivers are really shitty about not being consistent in the time they take to service the interrupt |
02:45.24 | tzanger | and jitter is the #1 killer of VOIP quality |
02:45.35 | tzanger | whether that jitter is network or system doesn't matter |
02:45.41 | Kb1_Kanob | Right, that also makes sense. |
02:46.04 | Kb1_Kanob | would kernel 2.6 provide better service to zaptel hardware? |
02:46.16 | tzanger | so for MOH timing and whatnot it is expecting that 1ms interrupt to occur with precision |
02:46.24 | tzanger | Kb1_Kanob: not sure, I don't use 2.6 for * |
02:46.52 | Kb1_Kanob | So it's just a trusted interrupt source, not a rolling counter in hardware. |
02:46.59 | tzanger | now SIP uses the incoming RTP stream to time (I call it self-timing) -- every time an RTP packet comes in the RTP clock advances... that's why VAD kills it at this time |
02:47.13 | tzanger | Kb1_Kanob: the zaptel hardware has this timer running and generating IRQs |
02:47.41 | tzanger | if the system doesn't respond to the IRQs in a timely fashion doesn't make any difference to the timer, it just keeps slapping the IRQ line |
02:48.01 | Kb1_Kanob | Hence setting the latency and priority would seem critical. |
02:48.24 | tzanger | I've seen that RTP (and IAX fromthe looks of it) timing is coming from an internal software timer now, or at least that is in the works -- the CVS logs showed some of that about a month back |
02:48.52 | Kb1_Kanob | this also explains the trunking/jitterbuffer incompatibility. |
02:48.55 | tzanger | so I believe that VAD and packet loss recovery will soon become features in * :-) |
02:49.07 | Kb1_Kanob | recovery = concealment? |
02:49.13 | tzanger | sorry yes concealment |
02:49.29 | tzanger | i.e. g729 and iLBC's PLC will become functional |
02:49.42 | Kb1_Kanob | This is fascinating - I have been seeking someone with understanding. |
02:49.50 | tzanger | right now they just never get called to do those functions because if packet doesn't come in, it doesn't get slapped to fake it |
02:49.58 | tzanger | Kb1_Kanob: I'm at the bottom end of the understanding |
02:50.08 | Kb1_Kanob | but you share. :-) |
02:50.14 | tzanger | kram or bkw or citats or coppice... those guys are really in the know |
02:50.38 | Kb1_Kanob | however, few seem to have the few minutes it takes at the moment. |
02:50.42 | Wi_Fi | they should be heading back home now |
02:50.43 | Wi_Fi | hehehe |
02:51.07 | Kb1_Kanob | tzanger: how are you with understanding the zap echo can? |
02:51.18 | tzanger | haha |
02:51.23 | tzanger | that is black juju, my friend |
02:51.29 | Wi_Fi | anyone in california? |
02:51.32 | tzanger | I don't think even the people who wrote it fully understand it |
02:51.34 | tzanger | :-) |
02:51.55 | Kb1_Kanob | I've been tring to understand where the taps occur and such. Been having a horrible local echo from my hybrid. |
02:52.14 | tzanger | it's kind of like the first Terminator -- the guys who built the skynet processors were given funky technology and told not to ask where it came from, but it gave them ideas and took them in directions they hadn't thought of :-) |
02:52.19 | Kb1_Kanob | Got an impedance check today from telco. Am getting 100ohm into a nominal 600ohm interface. |
02:52.23 | Kb1_Kanob | Not nice. |
02:52.29 | tzanger | Kb1_Kanob: is it balanced at least? |
02:52.42 | tzanger | take your ohmmeter and measure from tip to ground and ring to ground |
02:53.03 | Kb1_Kanob | Hmmm... I will check, however should be - have 11 trunks. |
02:53.11 | Kb1_Kanob | I am less than 500m from serving CO. |
02:53.12 | sudhir492 | Can anyone tell me why would registration fail in a SIP device, but the phone can still make calls through the server |
02:53.13 | tzanger | you can "balance" it by adding resistance to the low leg |
02:53.22 | tzanger | you're less than 500m and your impedance is fucked? |
02:53.29 | Kb1_Kanob | bottom of the scale. |
02:53.33 | tzanger | heh |
02:53.47 | tzanger | with 11 lines you're almost worth going to a PRI or cT1 |
02:54.01 | Kb1_Kanob | If I turn of the can the sidetine is horrible - almost as loud coming back as going out. |
02:54.06 | tzanger | just tell me you're using a channel bank :-) |
02:54.07 | Kb1_Kanob | PRI on order. |
02:54.28 | Kb1_Kanob | oh yeah - but it's got op amp hybrid and I can't twiddle to match down to 100ohm. |
02:54.30 | tzanger | and not 3 TDM404Ps |
02:54.44 | Kb1_Kanob | an old Newbridge Mainstreet 3624 |
02:54.50 | tzanger | hmm not familliar with that one |
02:54.53 | tzanger | funny thing is |
02:54.55 | tzanger | I am on PRI |
02:54.57 | tzanger | and I still get echo |
02:54.58 | Kb1_Kanob | mid 90s modular. |
02:55.13 | tzanger | my voice is bouncing off the far-end's hybrid and reflecting |
02:55.15 | Kb1_Kanob | oh? telco mismatch on their indermediate frame? |
02:55.22 | tzanger | but only for some numbers |
02:55.27 | Kb1_Kanob | Right, but it's fairly attenuated? |
02:55.41 | Kb1_Kanob | (if can is off?) |
02:55.45 | tzanger | most numbers are fine... but some aren't and it's not switch related |
02:55.51 | tzanger | no it's pretty loud actually :-) |
02:55.58 | Kb1_Kanob | Hmmm.... I had a report of one of those today. |
02:56.00 | tzanger | some it's barely noticeable but ohters it's distracting |
02:56.12 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
02:56.20 | tzanger | I've got the echo can on and set to 128 taps, played around with that a lot it's just not working right |
02:56.23 | Brixius | Hello |
02:56.27 | Kb1_Kanob | I'm assuming the can is synced to the near end sidetone and the far end echo was being ignored. |
02:56.43 | tzanger | probably, I can't say for sure |
02:57.00 | tzanger | I was going to buy a tellabs hardware 64ms echo can but I'm going ot try a few other things first |
02:57.11 | Kb1_Kanob | I can dial down to 32 taps w/o training and get 100% supression when I dial quiet termination test number, but in practice I need at least 128 w/ training=800 |
02:57.25 | file[laptop] | ARE YOU ON THE INTERWEB???!!! |
02:57.49 | Kb1_Kanob | taps are in 1 tap per sample, and one sample on t1 is 1/8000 of second, right? |
02:58.33 | tzanger | that's a good question I'm not sure but that certainly makes sense |
02:58.36 | Kb1_Kanob | so what are the units of the 'training' factor? |
02:58.42 | tzanger | Kb1_Kanob: taps :-) |
02:58.55 | Kb1_Kanob | Hmmm.... |
02:59.21 | tzanger | but yeah in a digital filter taps are usually discrete units of samples |
02:59.29 | Brixius | I just got a 7960 to replace a barbietone, that's like night and day |
02:59.33 | Wi_Fi | guys is it possible to have an iaxy be the default ring on incoming zap channel |
02:59.40 | tzanger | Brixius: yeah? |
02:59.40 | Kb1_Kanob | also makes sense, but training 'mutes' the channel to get the initial delay factor - however it can only mute the near end outbound, so it'll really try to compensate for far end echo initially.? |
03:00.13 | tzanger | Kb1_Kanob: it doesn't so much mute -- it mutes you off and sends an impulse down the line and measures the response to try and quickly train the echo canceller |
03:00.17 | tzanger | you can kill that by setting echotraining=off |
03:00.23 | tzanger | (or some # of ms) |
03:00.27 | tzanger | I think it defaults ot 800 |
03:01.11 | Kb1_Kanob | yes, 800 default. So it's really geared to supressing far end echo rather than excessive sidetone. |
03:01.53 | infinii | can someone show me their /etc/modprobe.d/zaptel file? I get errors on bootup |
03:01.56 | *** part/#asterisk invi_ (~undisclos@dsl-cap-209-5-169-185-cgy.nucleus.com) |
03:02.09 | Kb1_Kanob | Sidetone would have a delay of 1 or 2ms. |
03:02.23 | tzanger | Kb1_Kanob: that seems more of a txgain issue than echo |
03:02.28 | tzanger | but I really despise playing with gains |
03:02.49 | tzanger | sidetone = sounds like intermod in the old ham radio days? |
03:02.54 | Kb1_Kanob | yes - I've attenuated down at CB to help the can, but the sidetone is wicked loud. |
03:03.14 | Kb1_Kanob | Yes - perhaps we could squeeze 3 channels into 1 that way! |
03:03.19 | tzanger | hehehe |
03:04.15 | Kb1_Kanob | Any idea what the algorithmic implications of 'AGRESSIVE' supression are? |
03:04.30 | Kb1_Kanob | I understand it tends to mute higher frequencys. |
03:04.47 | tzanger | not sure there, youd' have to piss around in the source |
03:05.02 | Kb1_Kanob | Ok. |
03:05.27 | Kb1_Kanob | Thanks - I really appreciate your time discussing this. |
03:07.32 | _rcs | in sip.conf, what entry to you create so that when you refer to a fone from extensions.conf, you refer to it as a number and not an IP. eg 2000 |
03:07.36 | Phantom-X | http://bugs.gentoo.org/show_bug.cgi?id=66720 |
03:07.40 | _rcs | like with SIP/2000 |
03:08.47 | *** join/#asterisk NormAst (NormAst@Toronto-HSE-ppp3677931.sympatico.ca) |
03:09.31 | NormAst | f |
03:12.56 | Brixius | there is really good documentation as well as samples in the default sip.conf file for that, or on the wiki it's at http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
03:13.51 | Kb1_Kanob | tzanger: another question if I may - how much drive should a 0db signal (ie. a milliwatt test tone) represent on the zap channel? ztmonitor provides a notional representation of levels for conversation, but I'm unclear what to expect from a 0db source. |
03:15.30 | tzanger | that's actually a REALLY good question |
03:15.34 | tzanger | I wish I had a good answer for you |
03:15.42 | tzanger | I've only ever eyeballed it |
03:15.47 | tzanger | er... earballed I guess is the term |
03:15.52 | Kb1_Kanob | Same here. |
03:16.29 | Kb1_Kanob | In particular I noted that when dialed in to a milliwatt source provided by asterisk the zap gain adjust didn't seem to bring it down in ztmonitor but it did acoustically. |
03:16.39 | tzanger | hmm |
03:16.46 | tzanger | maybe the ztmonitor output is linear |
03:17.11 | Kb1_Kanob | Or the gain is inside the card and ztmonitor taps the bitstream. |
03:17.19 | *** join/#asterisk Mike (~mike@201.135.48.52) |
03:17.19 | tzanger | yes that'd do it too |
03:17.21 | Wi_Fi | can IVR be interupted? |
03:19.47 | pfn | what do you mean by interrupted |
03:20.37 | Wi_Fi | such as a phone that has PA when incoming calls ring and would like to listen to person calling then pick up in middle of leaving msg |
03:21.16 | Kb1_Kanob | Wi_Fi: are you describing 'caller screening' or 'recall from voicemail'? |
03:21.26 | Wi_Fi | caller screening |
03:21.27 | Wi_Fi | yes |
03:21.30 | Wi_Fi | thats the ticket |
03:21.44 | *** join/#asterisk lattera (~lattera@dialup-4.228.195.121.Dial1.Denver1.Level3.net) |
03:21.55 | lattera | anyone ever used asterisk on FreeBSD? |
03:21.56 | tzanger | anyway |
03:21.57 | tzanger | night all |
03:22.01 | Wi_Fi | tryin to make this transition as smooth as possible |
03:22.02 | florz | Any clue whether it is possible to find out the MSNs of a EuroISDN line using only the line and a computer connected to it? |
03:22.06 | Kb1_Kanob | tzanger: gnight & thanks again. |
03:22.12 | tzanger | np glad to help where I can |
03:22.18 | Wi_Fi | nite tzanger |
03:23.27 | *** join/#asterisk andy_b (~me@208-186-220-90.nrp4.brv.mn.frontiernet.net) |
03:23.43 | Wi_Fi | Kb1_Kanob is caller screening possible |
03:23.55 | lattera | anyone ever used asterisk on FreeBSD? |
03:23.57 | *** join/#asterisk chap (chapster@woodstock.cleburne.com) |
03:24.14 | andy_b | hello all - is anyone interested in helping a newbie tonight? (yeah yeah, I hear the groans...=)) |
03:24.24 | Wi_Fi | good |
03:24.24 | Wi_Fi | hehehe |
03:24.35 | Kb1_Kanob | Wi_Fi: honestly, I don't know. It could probably be done with zapbarge() if they were coming in over a zap channel. |
03:24.36 | Wi_Fi | caller screening on asterisk |
03:24.38 | Poemius | lattera : I tried using it 3 months ago, it did not compile, I hear they made progress since then, not sure |
03:24.41 | chap | Channel rules prohibit helping newbies. |
03:24.47 | andy_b | sweet |
03:24.49 | andy_b | =) |
03:25.00 | lattera | is zaptel _required_? |
03:25.09 | chap | :) |
03:25.10 | lattera | because zaptel does not compile |
03:25.22 | florz | chap: Can't you see that telling that a newbie helps him? |
03:25.40 | Wi_Fi | they are comming over zap channel |
03:25.58 | Poemius | lattera: you may need ztdummy for some functions, for music on hold to work |
03:26.02 | chap | florz: I see that now.. Damn my incompetence! |
03:26.04 | andy_b | I stumbled onto asterisk by accident actually - looks like a GREAT solution....I want to try it out - have it installed on a RedHat 9 box, but would like some guidance on initial setup...? |
03:26.10 | florz | chap: *gg* |
03:26.32 | Kb1_Kanob | Wi_Fi: yeah, try getting creative with zapbarge and the dialplan. |
03:26.40 | Wi_Fi | yikes |
03:26.49 | Wi_Fi | no guru here |
03:26.50 | Wi_Fi | hehe |
03:26.54 | andy_b | ;) |
03:26.54 | Kb1_Kanob | myself, I need 'recall transfered call' and/or 'recall from voicemail' in a PRI network. |
03:28.12 | andy_b | my goal for testing the basic functionality of this puppy out is to hook it up in my office - one analog phone line, one server, one handset |
03:28.24 | Connor_ | ~seen kram |
03:28.25 | jbot | kram is currently on #asterisk. Has said a total of 4 messages. Is idling for 5h 57m 20s |
03:28.41 | chap | andy_b: read, read, read. It is pretty cool stuff, I have only been messing with it for a few months- so my newbiew advice is to read. Don't forget about the wikki. |
03:28.49 | andy_b | wikki? |
03:29.06 | andy_b | I have the handbook from digium - going through that baby right now |
03:29.06 | Poemius | voip-info.org |
03:29.38 | Kb1_Kanob | andy_b: googling against digium.com and voip-info.com will prevent you pulling out most of your hair during your implementation. |
03:30.08 | andy_b | nice |
03:30.14 | andy_b | thanks all...=) |
03:30.32 | chap | anytime. Since I am rarely here. :) |
03:31.47 | Wi_Fi | Kb1_Kanob zapbarge in zapata.conf? |
03:32.34 | Kb1_Kanob | http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge |
03:35.32 | Wi_Fi | thnx bud |
03:37.10 | Kb1_Kanob | np |
03:38.20 | *** join/#asterisk drumkilla_ (~russelb@user-24-214-77-148.knology.net) |
03:38.20 | *** mode/#asterisk [+o drumkilla_] by ChanServ |
03:39.20 | JunK-Y | moooo |
03:39.46 | mishehu | The Kow says Mu. |
03:39.57 | Poemius | mooose? |
03:40.03 | *** join/#asterisk carlosmm (carlosm@adsl-66-142-143-244.dsl.hstntx.swbell.net) |
03:41.10 | carlosmm | Good Morning. |
03:41.38 | mishehu | Moose Poemius? |
03:43.53 | Poemius | lol |
03:43.54 | Poemius | I've had breakfast already, thank you :) |
03:45.02 | infinii | My 'ship show peers' shows my 2 klite phones logged on. however I can't call each other, I get 404 not found. Any ideas? |
03:45.13 | Kb1_Kanob | dial() to multiple zap targets - if one gives back a PRI busy then the dial falls through to the next step. How can I make it continue to timeout first (to ring other targets)? |
03:45.23 | Poemius | Moosehehu? :^) |
03:45.52 | *** join/#asterisk carlosmm (carlosm@adsl-66-142-143-244.dsl.hstntx.swbell.net) |
03:45.56 | carlosmm | Hola !! |
03:46.06 | Connor_ | checkout bug 2601 |
03:46.39 | enmaca | carlosmm: Hola !! |
03:47.12 | carlosmm | epale |
03:47.35 | *** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net) |
03:48.33 | carlosmm | NE1 knows how to make asterisk use MySQL or any other DB to autenticate and manage extensions ? |
03:48.49 | Poemius | check out the wiki |
03:48.57 | Q-At-Home | hows todays CVS |
03:49.15 | Q-At-Home | :) |
03:49.25 | *** join/#asterisk manipura (~mike@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
03:49.52 | pfn | q if you have to ask, use v1-0 |
03:50.05 | Q-At-Home | oh come on now, its what I ask every time I show up |
03:50.15 | Q-At-Home | its like... how I greet everyone :) |
03:50.50 | carlosmm | NE1 ? |
03:51.07 | pfn | carlosmm: |
03:51.09 | pfn | ~asterisk wiki |
03:51.10 | jbot | methinks asterisk wiki is http://www.voip-info.org/wiki-Asterisk |
03:51.15 | pfn | read and be enlightened |
03:51.23 | *** join/#asterisk plungerboy (meowmeow70@sdn-ap-003watacoP0336.dialsprint.net) |
03:51.24 | carlosmm | :D thanks i will |
03:51.47 | pfn | I hope the diamond I ordered comes out nice |
03:51.53 | Q-At-Home | so what have I missed during my 3 week absense |
03:52.26 | bkw_ | ack |
03:52.52 | Q-At-Home | hiya bkw |
03:57.10 | Q-At-Home | hrmm, does anyone know if theres a way to get the 2 port FXO to report when a line is unplugged ala red alarm? |
03:57.25 | Q-At-Home | or what the new way of telling if the line is gone is? |
03:57.27 | *** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com) |
03:57.32 | blankman | Hey guys. |
03:58.08 | plungerboy | yo |
04:01.49 | Q-At-Home | eh |
04:04.23 | blitzrage | we have an older Bell payphone in our classroom. I'm curious... could the tdm400p and the zaptel drivers have the ability to support the tones when you drop in the money? |
04:04.29 | pfn | 2 port fxo? |
04:04.38 | pfn | tdm400p won't go red |
04:05.14 | blitzrage | its plugged into a 2 port fxs |
04:05.31 | blitzrage | it works (I can ring the phone)... just curious about the coin mechanism tones |
04:06.02 | blankman | So I have a question ... what is the best media player on linux for the wav49 format? Is it the same as wav in the sense that all player should handle it? When I try to play it back with xmms it can't ... is there a plugin for it? |
04:07.36 | *** join/#asterisk InetNomad (~lenny@rockbox-gw.voiping.com) |
04:07.52 | Q-At-Home | hrm |
04:07.55 | ManxPower | "play" It comes with "sox" |
04:08.05 | InetNomad | Has anyone tried the parking and respective *8 parking pickup with chan_sip on 1.0.X? |
04:08.12 | Brixius | anyone know if there is a way to reboot a 7960 from the keypad, or do you just have to power cycle it |
04:08.14 | Q-At-Home | I've got a pair of tdm400p boards.. red was handy |
04:08.28 | Q-At-Home | hey ManxPower, long time no see :) |
04:08.32 | InetNomad | I keep getting "Oct 7 22:27:28 NOTICE[-170013776]: chan_sip.c:7284 handle_request: Nothing to pick up" |
04:09.08 | Q-At-Home | just waiting for the wife to get home before I compile tonites cvs... just in case |
04:10.24 | blitzrage | Brixius: try * 6 settings |
04:10.39 | blitzrage | Brixius: or telnet into it and do a 'reload' |
04:10.58 | Q-At-Home | isnt *8 for picking up a ring group? |
04:11.11 | Q-At-Home | (its been a while sinve I've been here so it may have changed) |
04:11.38 | InetNomad | Oh yea, you might be right. |
04:11.40 | InetNomad | Doh! |
04:11.57 | ManxPower | How do you do a factory reset on the SPA-3000 |
04:12.04 | Brixius | blitzrage: thanks |
04:12.07 | drumkilla_ | yep, it's for picking up a ring group |
04:12.12 | InetNomad | You dial the Pickup extension to get the extension... doh! |
04:12.14 | drumkilla_ | and by the way, that is configurable now to any extension |
04:12.24 | InetNomad | Thanks guys |
04:13.24 | infinii | woohoo!! ext to ext working internally. even voicemail works and it's sending an email with WAV attachment. |
04:13.31 | Brixius | now if I can only find the error in my dialplan.xml I'll be happy(er) |
04:18.42 | Juggie | can you guys take a look at this http://lists.digium.com/pipermail/asterisk-users/2004-October/066245.html and tell me what you think.... its really driving me insane. |
04:24.53 | pfn | so the first test would be to see what happens if you dial that number from a normal phone as opposed to your sip phone |
04:25.25 | pfn | you also have that timeout message |
04:25.38 | paulc | ~seen showtime |
04:25.40 | jbot | showtime <show@65.171.196.4> was last seen on IRC in channel #asterisk, 5d 3h 40m 47s ago, saying: 'dumb proxy and registrar'. |
04:27.10 | *** join/#asterisk kb1_kanob_ (johnsmith@sec2d48.dial.uniserve.ca) |
04:27.19 | infinii | hrm. voicemail works, but these attachments aren't meant for me to simply go into email and detach and listen to. I guess asterisk's voicemail system decodes it differently |
04:27.56 | infinii | anyways, I'm pleased to have internal crap working. I'll read more and explore later. g'nite |
04:31.53 | *** join/#asterisk Wi_Fi (~OUT@c-66-229-160-14.we.client2.attbi.com) |
04:35.11 | blankman | Hey ManxPower, so you are saying to use sox to "play" or to convert it to a new file type and play that? |
04:36.31 | Juggie | pfn, were you talking to me? there... sip phone works and i can hear the message, from outside the pri i only get fast busy after a few seconds, no audio |
04:36.48 | Juggie | but if i bridge a call to another sip client/server. it works |