| 00:00.10 | drocko | Maliuta: I'm using flowroute and they indicate that this should work |
| 00:00.14 | Maliuta | drocko: it's possible the issue is the SIP provider not letting you make the outgoing call |
| 00:00.18 | drocko | here is the error i see when i dial |
| 00:00.38 | drocko | or well, here are the messages i see |
| 00:00.47 | drocko | SIP/flowroute-081a24e0 is making progress passing it to SIP/68147638-0819cfa0 |
| 00:00.54 | drocko | SIP/flowroute-081a24e0 answered SIP/68147638-0819cfa0 |
| 00:01.01 | drocko | Attempting native bridge of SIP/68147638-0819cfa0 and SIP/flowroute-081a24e0 |
| 00:01.07 | *** join/#asterisk knarfly (n=knarfly@c-75-74-155-198.hsd1.fl.comcast.net) |
| 00:01.09 | Maliuta | learn to use pastebin |
| 00:01.19 | knarfly | where do I download the asterisk-gui |
| 00:01.19 | drocko | ok i can use that instead |
| 00:02.12 | Maliuta | knarfly: read the damn topic |
| 00:02.23 | drocko | Maliuta: here's the messages: http://pastebin.com/d33ae643 |
| 00:02.43 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 00:02.59 | *** part/#asterisk `paul (n=aldee@125.252.68.126) |
| 00:03.09 | Maliuta | drocko: and the dialplan? |
| 00:03.38 | Maliuta | drocko: well that and a SIP Debug might help |
| 00:03.40 | knarfly | Hey Maliuta take my advice and chill out dude...your ignorance is showing |
| 00:04.00 | Maliuta | knarfly: this is not the place for asterisk gui support |
| 00:04.06 | knarfly | Hey Maliuta take my advice and chill out dude...your ignorance is showing |
| 00:04.30 | drocko | Maliuta: here it is with the dialplan: http://pastebin.com/m27558559 |
| 00:05.58 | drocko | Maliuta: i've done a call with sip debug enabled |
| 00:06.48 | drocko | here's the sip debug: http://pastebin.com/d7f39b994 |
| 00:09.18 | drocko | i think maybe i'll just keep tinkering with it. |
| 00:09.28 | Maliuta | 192.168.200.99? no NAT? |
| 00:09.28 | drocko | i may also contact flowroute to see if the problem is with them |
| 00:09.41 | drocko | the asterisk box is natted |
| 00:09.49 | Maliuta | drocko: you can make calls on that peer normally? |
| 00:10.07 | drocko | Maliuta: i have used the same diaplan to dial US numbers without a problem |
| 00:10.18 | drocko | Maliuta: the only thing i've changed is the phone number to dial |
| 00:10.23 | Maliuta | drocko: with the current network configuration? |
| 00:10.45 | drocko | Maliuta: yes |
| 00:10.54 | drocko | Maliuta: that shouldn't work? |
| 00:11.34 | Maliuta | drocko: and you can call that sip number successfully from a handset/softphone attached to your *? |
| 00:12.14 | drocko | Maliuta: i have not tried that. what softphone do you reccomend? |
| 00:12.25 | Maliuta | drocko: i.e. calls to that extension don't fail under a circumstance that is not SIP->net->*->net->SIP |
| 00:12.35 | Maliuta | drocko: zoiper works |
| 00:12.58 | Maliuta | drocko: you don't have a hardphone/ata hooked up to the network the * box is on? |
| 00:13.08 | drocko | Maliuta: nope, just getting started with this guy |
| 00:13.32 | drocko | Maliuta: i guess you are right though, i'll work on getting it to work with a handset or a softphone and then try the dialplan |
| 00:13.36 | drocko | gtg now, thanks for the help! |
| 00:13.51 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 00:17.15 | eXistenZ | is it possible to connect skype and asterisk? |
| 00:17.40 | Maliuta | eXistenZ: I think chan_skype is not supported |
| 00:17.54 | Maliuta | eXistenZ: and it never was upto scratch |
| 00:18.07 | *** join/#asterisk Alton2 (n=alton@72.48.111.234) |
| 00:18.39 | eXistenZ | I see |
| 00:18.54 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:19.09 | eXistenZ | Maliuta, Is there a voip-service that gives unlimited landline calls in flat rate? |
| 00:19.32 | Maliuta | eXistenZ: depends on where you are I guess |
| 00:19.54 | eXistenZ | Maliuta, Israel |
| 00:20.03 | eXistenZ | I haven't found anyone other than skype |
| 00:20.12 | Maliuta | eXistenZ: I use one that gives me utimed calls to the US, UK, CA, HK, AU .... for $0.08AU/call |
| 00:20.38 | Maliuta | eXistenZ: you're looking to for landline calls in Israel? |
| 00:20.40 | JT | pennytel? |
| 00:20.41 | eXistenZ | Maliuta, Skype has now unlimited calls for like 3EU month |
| 00:20.58 | Maliuta | JT: yup, and I have my DID for the $10 Deposit |
| 00:21.03 | JT | heh |
| 00:21.05 | eXistenZ | Maliuta, I used voipbuster, it is supported as a trunk, but it is limited for like 136 free calls |
| 00:21.22 | eXistenZ | Maliuta, and pretty much cheap, 0,001⬠for min |
| 00:21.23 | JT | pennytel is pretty goo |
| 00:21.24 | JT | goo |
| 00:21.28 | JT | god damn it |
| 00:21.38 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:21.39 | eXistenZ | Why skype isn't supported |
| 00:21.49 | Maliuta | JT: quality is good, connectivity is good, price is good |
| 00:22.01 | JT | because it's a proprietary piece of crap protocol and program |
| 00:22.02 | Maliuta | JT: especially with no monthly |
| 00:22.07 | JT | skype is junk |
| 00:22.15 | JT | Maliuta: yeah |
| 00:22.20 | florz | eXistenZ: because they don't want to tell you how to speak to them - makes sense for a telephony system ... |
| 00:22.22 | eXistenZ | JT, but it has a good offer now |
| 00:22.45 | JT | eXistenZ: i couldn't care less really |
| 00:22.55 | florz | eXistenZ: no, technology that's constructed for enabling monopolies most likely is not a good offer |
| 00:23.29 | Maliuta | Skype is bad, not only is it crap "VoIP" and proprietary it's a massive security hole |
| 00:23.50 | florz | ... which they try to hide, too, obviously ... |
| 00:25.09 | Maliuta | I try to keep it out of any network I control |
| 00:35.17 | *** join/#asterisk seigneurpl (n=seigneur@206.167.57.180) |
| 00:35.50 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 00:38.09 | Robba | JT: do you know much about BRI? |
| 00:40.04 | pcrane | I need a hand with getting MySQL compiled in with asterisk addons |
| 00:40.29 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 00:41.09 | *** join/#asterisk profounded (i=d1b00403@gateway/web/ajax/mibbit.com/x-80a78d59a9855fd0) |
| 00:41.10 | pcrane | when I do make menuselect, it tells me that it can't add in mysql stuff as it depends on mysqlclient |
| 00:41.23 | pcrane | # find / -name mysqlclient |
| 00:41.23 | pcrane | # find / -name libmysqlclient.so |
| 00:41.30 | pcrane | <PROTECTED> |
| 00:41.30 | profounded | question, are there any pci express zaptel cards? |
| 00:41.51 | bad_duck | Hi, I'm a trying to install chan_mobile but after downloading asterisk-addons, I have an error while doing make menuselect : *** Install ncurses to use the menu interface! *** |
| 00:41.51 | bad_duck | - menuselect changes NOT saved! |
| 00:41.52 | bad_duck | But ncurses is installed (Asterisk 1.4 on debian etch) |
| 00:42.37 | Maliuta | profounded: check the digium site |
| 00:42.57 | bad_duck | I tried make distclean but no changes |
| 00:45.04 | TJNII | bad_duck: Did you re-run the configurator? I don't remember if re-running make would re run ./configure or not. |
| 00:45.04 | profounded | i should rather put it: are there any fxo:fso pci express port cards |
| 00:45.04 | bad_duck | TJNII: I tried make distclean, ./configure then make menuselect |
| 00:45.23 | TJNII | Okay, then I'm out of ideas. :) |
| 00:45.32 | bad_duck | :p thank you |
| 00:45.38 | Maliuta | profounded: the "zaptel" cards are all produced by digium go look there |
| 00:46.04 | TJNII | 's setup went smoothly and as such he forgot most of the steps. |
| 00:46.47 | JT | Robba: a bit, i've set it up before |
| 00:47.04 | *** join/#asterisk matt_keys (n=matt_key@h173.52.90.75.dynamic.ip.windstream.net) |
| 00:47.19 | *** join/#asterisk coolhp (n=Toshiba@modemcable157.57-80-70.mc.videotron.ca) |
| 00:48.45 | *** join/#asterisk UnixDawg (n=unixdawg@181.128.204.68.cfl.res.rr.com) |
| 00:48.58 | bad_duck | TJNII: it works with asterisk-addon-1.4.7 but not with the trunk version |
| 00:49.11 | UnixDawg | asterisk-now has moved to using freepbx gui ? |
| 00:49.24 | coolhp | Good evening everyone... Would anyone have tested the T.38 features in 1.6 ? I was wondering if they worked... |
| 00:50.24 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:51.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:51.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:51.39 | matt_keys | coolhp: enabling passthrough? |
| 00:52.13 | coolhp | matt_keys : No, T.38 origination and termination. |
| 00:52.28 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 00:53.14 | *** join/#asterisk astassistant (n=support@h-72-244-204-146.sttnwaho.dynamic.covad.net) |
| 00:53.37 | bad_duck | TJNII: I had an other version of asterisk-addon installed, it's maybe what makes troubles ? |
| 00:54.05 | TJNII | shrugs |
| 01:09.38 | *** join/#asterisk batcavejdt (n=batcavej@ip68-13-102-166.om.om.cox.net) |
| 01:11.19 | funjon | so, i'll ask again, since noone knew earlier, maybe someone else has an idea |
| 01:12.00 | funjon | so, i have an unusual hardware related question. I'm looking at replacing my dad's 25+ year old phone system (I think it's a Meridian) with an Asterisk box and some Cisco IP phones. That part I think I can handle, however, his office is 3-400 yards away from the actual shop (he owns an auto body repair shop - lots of noise in the shop). I need to implement a paging intercom speaker - a dedicated extension that plays a loud noise, then is a two-way spe |
| 01:13.14 | funjon | transit between buildings is no problem - wireless bridge + high gain yagi, i'm pretty sure we have line of sight. but implementing the intercom is about the only thing i haven't figured out yet |
| 01:14.19 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 01:15.11 | TJNII | Does it have to be a 2-way intercom? That seems like a bad idea in a loud shop. |
| 01:15.47 | funjon | yeah, it needs to be |
| 01:16.07 | funjon | they dont carry on long conversations, but it's usually "Hey, is $foo back there?" "yeah" "please call me at 08" or whatever |
| 01:16.11 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 01:18.26 | funjon | which, while noisy, is better than making someone drop their work (Because they'd never decide on who), go over, and call whatever extension, just to say no, $foo is gone |
| 01:19.31 | Robba | JT: Have you setup asterisk >1.4.18 with a B410P card? |
| 01:19.55 | *** join/#asterisk batcavejdt (n=batcavej@ip68-13-102-166.om.om.cox.net) |
| 01:20.40 | funjon | TJNII: of course, this is my dad, who is being dragged kicking and screaming into the 20th century (much less 21st). he's still on dialup and still uses a paper fax :P |
| 01:23.33 | JT | Robba: nah, i avoid that card really |
| 01:23.37 | JT | as it uses misdn |
| 01:24.05 | russellb | not for much longer, though |
| 01:25.10 | JT | oh? neat |
| 01:25.48 | florz | actually, it does already work with the bristuff multibri driver, IIRC |
| 01:26.01 | florz | at least with some patching |
| 01:26.23 | pcrane | I've got a problem with MySQL and Asterisk-Addons |
| 01:26.46 | pcrane | when I do ./configure this is what I get: |
| 01:26.47 | pcrane | checking for mysql_config... /usr/bin/mysql_config |
| 01:26.47 | pcrane | checking for mysql_init in -lmysqlclient... no |
| 01:27.05 | pcrane | and it doesn't allow me to select it in make menuselect |
| 01:27.36 | pcrane | # find / -name mysqlclient <nothing> |
| 01:27.41 | pcrane | # find / -name libmysqlclient.so |
| 01:27.47 | pcrane | <PROTECTED> |
| 01:27.48 | *** join/#asterisk bad_duck_ (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 01:27.52 | pcrane | any ideas? |
| 01:28.04 | pcrane | those are the standard places for mysql, right? |
| 01:28.09 | russellb | you probably don't have the -devel or -dev package installed ... |
| 01:28.18 | pcrane | I do |
| 01:28.33 | russellb | well, config.log will tell you what failed |
| 01:28.41 | pcrane | # find / -name mysql.h |
| 01:28.41 | pcrane | <PROTECTED> |
| 01:28.44 | pcrane | ok |
| 01:29.08 | pcrane | it says it was fine... |
| 01:29.08 | pcrane | configure: exit 0 |
| 01:29.24 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 01:29.24 | pcrane | <PROTECTED> |
| 01:29.26 | pcrane | hmmm |
| 01:29.31 | pcrane | how do I tell it where it is? |
| 01:29.39 | russellb | what does mysql_config --libs give you |
| 01:29.51 | pcrane | # mysql_config --libs |
| 01:29.52 | pcrane | -L/usr/lib64/mysql -lmysqlclient -lz -lcrypt -lnsl -lm -L/usr/lib64 -lssl -lcrypto |
| 01:29.58 | pcrane | hmm... |
| 01:30.01 | pcrane | that's interesting |
| 01:30.06 | russellb | well, there you go |
| 01:30.13 | pcrane | I think it should be /usr/lib/mysql... |
| 01:30.15 | russellb | asterisk is looking where it was told, and it's not there :) |
| 01:30.31 | pcrane | ok |
| 01:30.32 | pcrane | ... |
| 01:30.35 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 01:30.37 | pcrane | so, how do I change it? |
| 01:30.40 | russellb | shrugs |
| 01:30.53 | russellb | find out why mysql_config isn't giving you the right information |
| 01:31.04 | russellb | which is an issue specific to your distribution |
| 01:31.05 | pcrane | probably cause I put in a 32bit version in there... |
| 01:31.09 | pcrane | stupid centos |
| 01:31.15 | russellb | yes, centos is teh such |
| 01:31.18 | russellb | suck* |
| 01:31.19 | pcrane | cheers russellb |
| 01:31.21 | pcrane | mmm |
| 01:31.24 | pcrane | very much so |
| 01:31.26 | russellb | :) |
| 01:31.30 | pcrane | gimme debian anyday |
| 01:35.14 | *** join/#asterisk Merch^ (i=merchant@sky.net.ai) |
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| 01:41.39 | *** part/#asterisk Paige (n=Paige@2001:470:b:aa:21f:c6ff:fe48:8ce3) |
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| 01:45.52 | *** join/#asterisk edwin_quijada (i=edwin_qu@190.166.56.30) |
| 01:46.08 | funjon | ah well, i'll email asterisk-users@ and see if anyone has any ideas on implementing an intercom |
| 01:46.17 | funjon | or ask the almighty lazyweb |
| 01:47.20 | Qwell | pcrane: topic :D |
| 01:47.33 | Qwell | ~centos52 |
| 01:47.41 | Qwell | ~centos52bug |
| 01:47.41 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 01:48.08 | pcrane | hmm... |
| 01:48.22 | pcrane | so, you're saying that I'm using the wrong arch? |
| 01:48.24 | pcrane | *sigh* |
| 01:48.39 | Qwell | it's stupid with deps, you need to be explicit and `yum install package.x86_64` or it'll break things |
| 01:48.41 | Qwell | quite badly |
| 01:48.51 | pcrane | I can't use yum |
| 01:48.59 | pcrane | the machine doesn't have direct access to the internet |
| 01:49.09 | Qwell | ahh, so you might be okay |
| 01:49.18 | Qwell | just install the same packages but x86_64 |
| 01:49.32 | pcrane | then I have to do the whole dependance resolution myself |
| 01:49.34 | pcrane | *sigh* |
| 01:49.45 | Qwell | well, if you just get the same packages you got last time.. the deps wouldn't change |
| 01:49.52 | pcrane | mmm |
| 01:49.54 | pcrane | I know |
| 01:49.57 | Qwell | well, shouldn't |
| 01:50.06 | Qwell | yeah, it's a PITA. Go yell at redhat ;/ |
| 01:50.18 | pcrane | well |
| 01:50.34 | pcrane | if I had internet access direct, then it'd not be a problem |
| 01:50.37 | pcrane | yum install stuff |
| 01:50.45 | pcrane | but I have to do rpm -i package 1 |
| 01:50.57 | pcrane | then it fails cause it needs packages 2 & 3 |
| 01:51.05 | pcrane | package 2 fails cause it needs package 4 |
| 01:51.06 | pcrane | and so on |
| 01:51.11 | pcrane | it's a mission |
| 01:59.37 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 01:59.38 | *** join/#asterisk aarcane (n=aarcane@c-24-7-144-248.hsd1.ca.comcast.net) |
| 01:59.43 | drmessano | AsteriskNOW is going to include FreePBX? |
| 02:00.04 | Qwell | That's what I've been hearing |
| 02:00.09 | aarcane | okie, so.. I have one of them thar fancy new dell laptops with a conexant modem. |
| 02:00.16 | Qwell | aarcane: no you can't |
| 02:00.19 | Qwell | don't even ask :) |
| 02:00.23 | aarcane | o,.,0 |
| 02:00.33 | aarcane | wow |
| 02:00.36 | Qwell | You can't connect a phone line to it, and use Asterisk with it. |
| 02:00.38 | Qwell | it won't work |
| 02:00.56 | aarcane | can you give me a short version of why not ? |
| 02:01.12 | drmessano | bcuz |
| 02:01.17 | drmessano | Thats 4 letters |
| 02:01.20 | Qwell | You'll need a piece of real telecom hardware, either an ATA, or a PCI card *not* a modem |
| 02:02.04 | aarcane | I'm pretty sure it's one of them fancy voice modems.. if it IS a voice modem, will it work ? (I can find some way to confirm..) |
| 02:02.07 | Qwell | no |
| 02:02.14 | drmessano | no |
| 02:02.15 | Qwell | well, yes |
| 02:02.22 | Qwell | if you can write a driver for it |
| 02:02.30 | Qwell | if you do - more power to you |
| 02:02.36 | drmessano | Hey Qwell |
| 02:02.44 | aarcane | I don't expect miracles, I just want enough functionality to experiment and learn how asterisk works. |
| 02:02.45 | Qwell | but really..no. it's just not worth it |
| 02:02.56 | Qwell | aarcane: TDM410 with 1 FXO module. |
| 02:03.10 | Qwell | hey drmessano |
| 02:03.35 | drmessano | Remember that guy that came in here the other day.... and he was like "I WANT MY MODEM TO WORK WITH ASTERISK" and we were like "No, theres no drivers" and he was like "FINE! IMMAGO MAKE ONE!! *HRMPH*" and ran off? |
| 02:03.43 | drmessano | Did he ever come back? |
| 02:03.46 | Qwell | no |
| 02:03.55 | aarcane | hrrm. |
| 02:04.05 | aarcane | does it need an asterisk driver, or a linux kernel driver ? |
| 02:04.22 | aarcane | I know it's on an HDA INtel Audio bus.. so it's connected to a sound card. |
| 02:04.26 | Qwell | aarcane: Linux kernel driver (specifically, a Zaptel driver) |
| 02:04.40 | Qwell | aarcane: It's really just not feasible |
| 02:05.09 | Qwell | drmessano: it's actually happened a few times |
| 02:06.24 | aarcane | Qwell, so to be able to make a phone call using asterisk on my laptop, I would have to install a pci bus and a fancy $544.39 or higher modem/sound card in one ? |
| 02:06.46 | Qwell | aarcane: it's significantly less than that, but, on a laptop, you could just use an ATA |
| 02:06.52 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 02:06.57 | aarcane | what's an ATA ? |
| 02:06.59 | Qwell | a linksys is like...I don't know, $80-100? |
| 02:07.01 | Qwell | ~ata |
| 02:07.01 | jbot | ata is probably Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 02:07.05 | drmessano | Why the hell would you install Asterisk on a laptop? |
| 02:07.18 | Qwell | You need an ATA with an FXO, to connect a phone line (there are very few of those - I think Linksys makes one) |
| 02:07.26 | drmessano | SPA-3102 |
| 02:07.31 | Qwell | that's the one |
| 02:07.35 | *** join/#asterisk dwayne (i=dwayne@76.29.245.9) |
| 02:07.35 | edwin_quijada | hi |
| 02:07.41 | aarcane | drmessano, like I said before, I want to experiment with it, and get to know the inner workings without spending a fortune. |
| 02:07.41 | Qwell | right dwayne ? |
| 02:07.59 | dwayne | Qwell: of course |
| 02:08.06 | drmessano | aarcane: You wont be installing real hardware on a laptop |
| 02:08.15 | drmessano | Like Zaptel cards you'll actually run into in the field |
| 02:08.27 | Qwell | I suppose you could use a Xorcom USB thingie... |
| 02:08.37 | Qwell | I don't know how small/expensive those are though |
| 02:08.43 | Qwell | best off getting an ATA |
| 02:09.12 | aarcane | hrrm |
| 02:09.13 | jaytee | drmessano, the O'Reilly book VOIP Hacks shows you how to hack the code to make a winmodem work like an fxs fxo (snicker) |
| 02:09.49 | Qwell | jaytee: good luck finding chan_modem |
| 02:09.52 | drmessano | Any book that claims to show you "hacks" is a fucking hack |
| 02:09.54 | aarcane | alright then. I guess I'm not ready for asterisk yet. I'll be back with I have some money to plunk into it :) |
| 02:10.01 | Qwell | aarcane: of course... |
| 02:10.05 | drmessano | bye |
| 02:10.06 | mosty | i'm trying to build app_meetme.so (and app_page.so) with asterisk 1.2.30, i have ztdummy loaded but there's no error and these modules aren't being built. what else do i need/where can i look to see why they aren't being built? |
| 02:10.07 | Qwell | you don't HAVE to connect to an analog line |
| 02:10.09 | jaytee | Qwell, no it's a hack to zaptel for the tdm driver |
| 02:10.11 | drmessano | See you on Skype |
| 02:10.13 | Qwell | nothing stopping you from using SIP |
| 02:10.14 | *** join/#asterisk JenniferAkemi (n=akemi@MTLXPQAK-1178074603.sdsl.bell.ca) |
| 02:10.19 | aarcane | Qwell, SIP ? |
| 02:10.25 | Qwell | jaytee: yeah...I don't buy it |
| 02:10.30 | Qwell | aarcane: yes, here |
| 02:10.31 | Qwell | ~book |
| 02:10.31 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:10.42 | aarcane | I have a comcast modem/phone line I was going to try to connect to the phone line and make a few calls back and forth. |
| 02:10.44 | jaytee | Qwell, yeah I wouldn't waste my time either, hence the snickering |
| 02:10.55 | drmessano | Notice how everything is a "hack" nowadays |
| 02:10.57 | drmessano | Like |
| 02:11.08 | Qwell | aarcane: get an account with an ITSP, it'll cost you a few bucks to get that going |
| 02:11.12 | Qwell | then you can use a softphone on your laptop |
| 02:11.13 | drmessano | "COOL TOP SECRET CHEESE SANDWICH HACKS YOUR DELI DOESNT WANT YOU TO KNOW" |
| 02:11.20 | jaytee | drmessano, didn't buy the book, was just leafing through it at the bookstore |
| 02:11.23 | Qwell | aarcane: check out the book - it'll get you started |
| 02:12.41 | drmessano | How to HACK your TOASTER for HACKED warm bread |
| 02:13.06 | drmessano | 20 ways to HACK a soda can to make the top open FASTER |
| 02:13.37 | drmessano | I bet I could write a book that turns taking a shit into a MAD HACK |
| 02:13.38 | jeev | mr qwell |
| 02:13.43 | Qwell | mr jeev |
| 02:13.48 | *** join/#asterisk JenniferAkemi- (n=akemi@MTLXPQAK-1178074603.sdsl.bell.ca) |
| 02:13.59 | jaytee | only voip book other than "The Book" that I was really interested in never made it to print, "Asterisk Cookbook". evidently lbadsen is too busy on the 3rd edition of TFOT or there's not enough demand for the title in advance. |
| 02:14.02 | jeev | our driver in washington d.c. kept calling me by my first name.. mr ... |
| 02:14.14 | jeev | l badsen?!1 |
| 02:14.30 | jaytee | lmadsen |
| 02:14.47 | jaytee | what're ya doin in D.C.? |
| 02:14.54 | unpaidbill | what is the the term for redirecting a call to another # so it isnt going through my pbx? |
| 02:14.55 | jeev | i was in d.c. for africa rising 2008 |
| 02:15.00 | unpaidbill | it's on the tip of my tongue |
| 02:15.02 | unpaidbill | argg |
| 02:15.05 | jeev | friend was in the show, he took me with him |
| 02:15.20 | Qwell | unpaidbill: so what isn't? |
| 02:15.30 | jeev | what're, wow. what were, never seen anyone say that before. |
| 02:15.31 | Qwell | like on a PRI? |
| 02:15.36 | Qwell | 2BCT? |
| 02:15.42 | unpaidbill | 2BCT yes |
| 02:15.45 | unpaidbill | thanks |
| 02:15.57 | Qwell | I'm disturbed that I know that |
| 02:16.15 | unpaidbill | yeah 2bchannel transfer unf |
| 02:16.18 | unpaidbill | haha im glad you did |
| 02:16.59 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-241-147.balt.east.verizon.net) |
| 02:19.04 | drmessano | MAD ASTERISK HACKS THE PHONE COMPANY DOESNT WANT YOU TO KNOW |
| 02:19.13 | unpaidbill | i'd buy it. |
| 02:22.39 | drmessano | IPHONE HACKING WITH ASTERISK BY RON PAUL <--- Best Seller |
| 02:22.48 | tzanger | ha |
| 02:23.16 | edwin_quijada | i get this error from asterisk and dmesg rtc: lost some interrupts at 1024Hz |
| 02:23.25 | edwin_quijada | sombody has any idea |
| 02:23.43 | mosty | edwin_quijada, is it a dell machine, and are you using ztdummy? |
| 02:23.44 | drmessano | Nope |
| 02:24.14 | edwin_quijada | mosty: Yeah!! |
| 02:24.56 | mosty | edwin_quijada, upgrade your kernel to a recent 2.6 kernel with hrtimers enabled, and use the latest 1.4 version of zaptel |
| 02:25.08 | mosty | or just don't use ztdummy |
| 02:25.17 | mosty | or don't use a dell |
| 02:25.54 | edwin_quijada | i dont have zaptel cards so how can i get the timer? |
| 02:26.22 | mosty | <mosty> edwin_quijada, upgrade your kernel to a recent 2.6 kernel with hrtimers enabled, and use the latest 1.4 version of zaptel |
| 02:26.56 | edwin_quijada | i cant upgrade to 2.6.18 kernel above |
| 02:27.05 | edwin_quijada | :( |
| 02:27.12 | mosty | edwin_quijada, it won't work then |
| 02:27.58 | mosty | maybe you can buy a cheap zaptel clone card on ebay and install that just for timing |
| 02:32.54 | edwin_quijada | mosty: check this! When I run the command genzaptelconf -sdvM |
| 02:33.03 | edwin_quijada | I fix the problem! |
| 02:33.25 | edwin_quijada | but when I reboot my machine I get the same |
| 02:34.16 | Steve_J-obs | Hi guys!: question: How do I send a sip register request from the dialplan..or alternate solution? |
| 02:36.34 | [TK]D-Fender | Steve_J-obs: You don't |
| 02:36.54 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:36.58 | [TK]D-Fender | Steve_J-obs: * will register on it own, and at the interval specified by your peer. |
| 02:37.52 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:38.02 | Steve_J-obs | yes, but in this circumstances, asterisk is already running, and I cant stop it to put a register command on the sip.conf |
| 02:38.36 | [TK]D-Fender | Steve_J-obs: You don't HAVE to stop * to take a change in sip.conf |
| 02:38.40 | Steve_J-obs | I am thinking maybe I can send a register request from a routine outside asterisk? |
| 02:38.54 | [TK]D-Fender | Steve_J-obs: No. |
| 02:38.58 | [TK]D-Fender | Steve_J-obs: see above |
| 02:39.16 | Steve_J-obs | tk-d-fender: That is a good idea... I had not thought of that |
| 02:39.35 | Steve_J-obs | wow |
| 02:39.48 | Steve_J-obs | tk-d-fender, you are a genious |
| 02:39.57 | [TK]D-Fender | Steve_J-obs: Next time just ask for what you'd like to do, not validation on each broken appoach YOU can come up with :) Its far less limiting... |
| 02:41.21 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-877c63ef1b79491b) |
| 02:41.45 | Steve_J-obs | I will like to write to sip.conf from the dialplan... although kind of tricky... there should be a couple of ways to do that... |
| 02:44.04 | batcavejdt | does asterisk now support cepstral out of the box- I know you have to buy some licenses - and I don't see "say text" as an option |
| 02:44.11 | [TK]D-Fender | Steve_J-obs: your concept is quite broken indeed. |
| 02:44.46 | [TK]D-Fender | Steve_J-obs: Why on earth would you be changing peers & the line from the DIALPLAN? |
| 02:45.20 | [TK]D-Fender | batcavejdt: Did you compile and * app that uses Cepstral? |
| 02:45.30 | [TK]D-Fender | batcavejdt: Like I dunno... app_cepstral? |
| 02:46.00 | Steve_J-obs | ok: I am writing an ivr, where people can call in an enter their dids... the ivr then will register the dids with another server |
| 02:46.03 | batcavejdt | I am new to this - I just installed the asterisknow and I saw it installs some cepstral apps :) |
| 02:46.44 | [TK]D-Fender | batcavejdt: Get reading then, you may be close. |
| 02:47.14 | Steve_J-obs | since the register command can only be on sip.conf, it will require that the ivr(dialplan) appends it to sip.conf |
| 02:47.33 | batcavejdt | what I am trying to do is use the telnet inteface to dial (using vb.net threads) and then route those to a voice prompt queue where I could say some text |
| 02:47.37 | [TK]D-Fender | Steve_J-obs: you are throwing "DID" around like its an all inclusive magical term. An who configures account info like this via IVR? |
| 02:47.50 | Steve_J-obs | but you gave me the idea my friend |
| 02:48.33 | [TK]D-Fender | Steve_J-obs: How I could possibly have inspired you to come to that end I desire no credit for. |
| 02:49.03 | Steve_J-obs | well, you did...you get the credit...thanks... I appreciate your help |
| 02:50.02 | [TK]D-Fender | backs away slowly.... |
| 02:51.24 | Steve_J-obs | tk-d-fender: do not mind.. it is 100% legitimate use |
| 02:51.46 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 02:52.05 | nny_1 | can a system use realtime for certain contexts in the dialplan only? |
| 02:52.45 | [TK]D-Fender | Steve_J-obs: I never though it was "not legitimate", its that I thinkk your concept is just plain WRONG. Just because you CAN do a thing doesn't mean you SHOULD |
| 02:52.59 | [TK]D-Fender | nny_1: Yes, thats how the realtime switch works. |
| 02:53.18 | [TK]D-Fender | nny_1: It REQUIRES extensions.conf to define which contexts will look outside |
| 02:53.24 | nny_1 | [TK]D-Fender: awesome |
| 02:53.35 | nny_1 | [TK]D-Fender: working on an interface to add users and vm |
| 02:53.39 | nny_1 | almost done actually |
| 02:55.46 | nny_1 | i like the idea of adding users/ defining which macros they use, voicemail etc etc in a DB, but loathe the idea of actual nuts and bolts of the Dp being used in realtime. I hope this is not a bad thought |
| 02:56.42 | nny_1 | hope that makes sense |
| 03:00.45 | brodiem | [TK]D-Fender: You may know this.. is the DSP hardware and specifically audio/speakerphone performance the same between an IP330 and IP501? |
| 03:01.06 | brodiem | nny_1: meh both have their pros and cons |
| 03:01.52 | nny_1 | brodiem: understood |
| 03:02.13 | nny_1 | we are gonna hybrid it so there is some easy way to add a user, without making trixbox etc etc etc |
| 03:02.51 | nny_1 | already have a MOH uploader with a sox script and some other simple stuff |
| 03:04.15 | brodiem | nny_1: yeah if using a gui wrapper to manage your users/exts then realtime is the way to go IMO |
| 03:04.22 | [TK]D-Fender | brodiem: Don't know. I hear the 320/330 is god, but I haven't used it personally. |
| 03:04.32 | nny_1 | brodiem: yeah |
| 03:04.42 | nny_1 | ok time to watch a movie later all thanks for the input |
| 03:06.12 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 03:06.35 | brodiem | [TK]D-Fender: heh I have an IP330 but at a location where I never use it on speaker. |
| 03:06.55 | brodiem | Hasn't struck me as god-like yet though lol |
| 03:07.26 | [TK]D-Fender | brodiem: its an IP 330... it isn't supposed to be god-like. IP 6XX is where its at :) |
| 03:08.56 | brodiem | [TK]D-Fender: yeah that was part of my thought also - i.e. Does a 650 give you anything more than a 330 besides extra lines, bigger display, USB storage and other non-hardware advantage? |
| 03:11.41 | brodiem | and g722 of course |
| 03:12.24 | [TK]D-Fender | brodiem: G.722, RJ9 headset, better speakerphone, more soft-keys. |
| 03:12.37 | [TK]D-Fender | brodiem: and backlight |
| 03:12.41 | brodiem | hmm so the speakerphone is improved |
| 03:12.48 | [TK]D-Fender | brodiem: and the ability to add attendant modules. |
| 03:13.04 | [TK]D-Fender | brodiem: for sure tis gotta be better, jsut thing of the acoustics of the base. |
| 03:13.30 | [TK]D-Fender | brodiem: Also has lit SPKR / Mute, etc |
| 03:14.06 | brodiem | [TK]D-Fender: What do you mean by the acoustics of the base? |
| 03:14.36 | [TK]D-Fender | brodiem: the 650 is physically larger and for the speaker in it would resonate better. Just a matter of build size, etc |
| 03:16.58 | *** part/#asterisk batcavejdt (n=batcavej@ip68-13-102-166.om.om.cox.net) |
| 03:17.07 | brodiem | I heard that the speaker is in a sealed enclosure, although not from a reliable source, but if it were true then it should be the same enclosure size |
| 03:17.33 | *** join/#asterisk elux (n=pak@CPE001ee5344dde-CM0018c0b38594.cpe.net.cable.rogers.com) |
| 03:17.35 | elux | hey guys |
| 03:17.40 | brodiem | well I suppose there's only one way of truly knowing the difference |
| 03:18.00 | elux | ive been hearing a lot about this project for years.. one thing i can quite understand is with it, i can essentially become a voip provider, .. ? |
| 03:18.12 | elux | as in, i can become my own voip provider to make calls |
| 03:18.42 | [TK]D-Fender | brodiem: Thing of the overall mass of the phones by comparison. the 6XX is much larger |
| 03:18.56 | *** join/#asterisk elux (n=pak@CPE001ee5344dde-CM0018c0b38594.cpe.net.cable.rogers.com) |
| 03:18.58 | elux | .. |
| 03:19.04 | [TK]D-Fender | elux: Yes. |
| 03:19.14 | elux | what i dont quite understand is what is the bridge between a traditional telephony system and asterisk? |
| 03:19.22 | elux | i mean, what bridges the calls, incoming and outgoing? |
| 03:19.34 | [TK]D-Fender | elux: * can be a lot of different things. You can build a PBX out of it. In that sense you could use it to provide services to outside clients, etc. |
| 03:19.36 | elux | there must be some service i must connect to and pay for per minute, or on some usage |
| 03:19.54 | [TK]D-Fender | elux: You could make a dating service, polling station, remote coffee-maker, just about anything. |
| 03:20.09 | elux | remote coffee-maker triggered via the phone? |
| 03:20.13 | elux | thats pretty incredible. |
| 03:20.20 | [TK]D-Fender | elux: there are plenty of PSTN service providers out there. These are called ITSPs |
| 03:20.22 | [TK]D-Fender | ~itsp |
| 03:20.23 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:20.29 | elux | any books you can recommend for me to get started on getting this understood? or other resources? |
| 03:20.39 | [TK]D-Fender | ~book |
| 03:20.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:20.41 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 03:20.48 | [TK]D-Fender | ~wikis |
| 03:20.48 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 03:20.50 | elux | does that talk abot ITSPs ? |
| 03:20.53 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 03:20.57 | elux | the book that is |
| 03:21.16 | elux | ~itsplist-ca |
| 03:21.17 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 03:21.40 | elux | how does a company like unlimitel offer its service? |
| 03:21.55 | elux | what technology/protocol do they use? |
| 03:22.02 | elux | just so understand the entire picture |
| 03:22.05 | mosty | perhaps you should ask them? |
| 03:22.24 | [TK]D-Fender | elux: SIP |
| 03:22.31 | elux | i see |
| 03:22.49 | [TK]D-Fender | elux: Go read the book and learn about the different technologies * can interact with |
| 03:22.56 | elux | ill order it asap |
| 03:23.16 | JunK-Y | its unlimitel.ca, not .com :) |
| 03:23.32 | elux | ok that makes more sense |
| 03:23.34 | [TK]D-Fender | JunK-Y: Salut mon ostie! |
| 03:23.40 | JunK-Y | [TK]D-Fender: yo! |
| 03:23.43 | elux | i was just on the .com and it didnt look right |
| 03:23.44 | JunK-Y | whats up? |
| 03:24.42 | [TK]D-Fender | JunK-Y: All sorts. Getting into protography and made a GIANT score on some Minolta gear today. |
| 03:24.57 | [TK]D-Fender | JunK-Y: http://forums.dpreview.com/forums/readflat.asp?forum=1037&thread=29076245 |
| 03:25.29 | [TK]D-Fender | JunK-Y: just ONE of those lenses is worth over 200$ easy |
| 03:27.17 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.143) |
| 03:27.33 | brodiem | [TK]D-Fender: FYI I just noticed on their data sheet, the list 320/330's DSP as "Acoustic Clarity", and the 550/650 as "Acoustic Clarity 2" - could just be the addition of G722 though I guess.. |
| 03:28.09 | [TK]D-Fender | brodiem: I discard teminology like that with extreme prejudice :) |
| 03:28.17 | brodiem | haha |
| 03:28.18 | elux | ~itsplist-us |
| 03:28.18 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 03:28.18 | [TK]D-Fender | brodiem: Worthless wind |
| 03:28.45 | brodiem | yea, time to stop theorizing |
| 03:29.07 | JunK-Y | [TK]D-Fender: nice, i feel like cheap with my old canon 7.1 :) |
| 03:29.21 | jeev | FENDERRR! |
| 03:30.24 | [TK]D-Fender | JunK-Y: I've gone DSL with the Sony A200. Was an awesome buy at $480 for the base & kit lens |
| 03:30.56 | [TK]D-Fender | JunK-Y: Everything else has been getting old Konica/Minolta lenses via Cragislist, Kijiji, etc |
| 03:32.19 | mchou | hey, anyone have any experience with future-nine as ITSP? Especially regarding CNAM (inbound caller ID)? |
| 03:32.29 | JunK-Y | kk |
| 03:33.48 | *** join/#asterisk Johnakabean (n=Johnakab@pool-72-82-111-106.nrflva.east.verizon.net) |
| 03:33.56 | mchou | some forum postings seem to indicate the Future-nine has reliable CNAM..... |
| 03:37.07 | *** join/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 03:37.26 | *** join/#asterisk Levonk (n=lk@adsl-75-62-134-101.dsl.lsan03.sbcglobal.net) |
| 03:37.38 | *** join/#asterisk xenonex (n=xenonex@82.200.211.39) |
| 03:37.58 | *** join/#asterisk marklynch (n=markl@226.10.220.203.static.comindico.com.au) |
| 03:40.15 | HumanCell | Question ... I'm not getting the CallerID that I want ... using the following: |
| 03:40.33 | HumanCell | exten=s,1,NoOp(RINGGROUP) |
| 03:40.33 | HumanCell | exten=s,2,Set(CALLERID(all)=Distinguished <5556080269>) |
| 03:40.33 | HumanCell | exten=s,n,Dial(SIP/5000&IAX2/5000&SIP/15552222509@trunk_3,20) |
| 03:41.39 | HumanCell | When Asterisk dials out to the third SIP number (which I changed for the example) there is no CallerID being set ... I'm I doing something obviously studpid? |
| 03:42.11 | HumanCell | thinking how stupid to not spell stupid correctly ... |
| 03:42.31 | mosty | HumanCell, does the "trunk_3" provider allow you to set arbitrary caller id? |
| 03:43.00 | HumanCell | I believe I have done it before ... it's VoicePulse ... |
| 03:43.42 | mosty | you should ask them |
| 03:44.00 | baliktad | I set callerid with voicepulse |
| 03:44.20 | mchou | HumanCell: You're saying Dial 2 resources the caller ID doesnt set, but dialing one resource will? |
| 03:45.08 | HumanCell | let me check something here ... |
| 03:45.46 | baliktad | I just use a normal Set(CALLERID(number)=5555551234) |
| 03:46.34 | HumanCell | baliktad: yes ... I wanted to set both with ALL ... but I'm going to try just number ... |
| 03:47.03 | baliktad | i haven't found a voip provider yet that passes name |
| 03:47.16 | baliktad | or more importantly, an endpoint that accepts one |
| 03:47.32 | baliktad | usually the local carriers are just doing name lookups in their own database |
| 03:47.37 | mchou | baliktad: yeah, that sucks |
| 03:48.03 | mchou | baliktad: that's one deficiency with voip |
| 03:48.10 | elux | hey guys, are the ITSPs listed in ~itsplist-ca and us friendly to hooking up a PBX to their network? |
| 03:48.21 | [TK]D-Fender | elux: Yes |
| 03:48.26 | elux | awesome |
| 03:48.27 | elux | thx |
| 03:48.30 | [TK]D-Fender | elux: They're there for a reason |
| 03:48.39 | elux | makes sense. just wanted to double check |
| 03:49.07 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.93) |
| 03:54.30 | mosty | what are the build dependencies for app_meeme.so in asterisk 1.2.30? i can't seem to get it to build, and i have zaptel installed and ztdummy loaded |
| 03:55.22 | [TK]D-Fender | mosty: And you compiled * AFTER loading ztdummy? |
| 03:55.36 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:55.36 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:55.36 | [TK]D-Fender | mosty: and runninh "ztcfg -vvvv" prior to starting *? |
| 03:55.37 | mosty | [TK]D-Fender, yes |
| 03:55.48 | [TK]D-Fender | mosty: pastebin "ztcfg -vvvv" |
| 03:56.04 | [TK]D-Fender | mosty: and then "load app_meetme.so" and "zap show status" |
| 03:56.31 | mosty | i hadn't run ztcfg, trying that now |
| 03:56.33 | [TK]D-Fender | mosty: And while your'e at it "load chan_zap.so" |
| 03:56.35 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 03:56.35 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:56.49 | [TK]D-Fender | mosty: amazing.. it was a "yes" just a moment ago.. |
| 03:57.41 | mosty | i hit enter before you mentioned ztcfg, or at least before it arrived on my screen here |
| 03:59.13 | mosty | ok, after running ztcfg, neither chan_zap.so nor app_meetme.so are built |
| 03:59.52 | [TK]D-Fender | mosty: Where's my pastebin backup of everything I asked you for? |
| 04:01.11 | jeev | LEIF |
| 04:02.20 | mosty | [TK]D-Fender, ztcfg -vvvv says no channels configured, and mentions the version (1.4.10.1), and none of the asterisk commands work because there is no chan_zap.so to load |
| 04:03.07 | [TK]D-Fender | mosty: ..... Zaptel 1.4 doesn't WORK with * 1.2!!! |
| 04:03.11 | [TK]D-Fender | mosty: Do you use diesel engine parts in your ELECTRIC car? |
| 04:03.23 | mosty | [TK]D-Fender, tzafrir said that is does |
| 04:03.35 | [TK]D-Fender | mosty: Put. Down. The. Crack. Pipe. (c) JerJer |
| 04:03.40 | [TK]D-Fender | mosty: FFS no. |
| 04:03.54 | [TK]D-Fender | mosty: No go compile a PROPER version. |
| 04:03.57 | [TK]D-Fender | now* |
| 04:07.38 | *** join/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 04:10.25 | JunK-Y | [TK]D-Fender: do you know how to see resources taken on the kernel land? |
| 04:10.44 | HumanCell | baliktad: So I have tested ... even just using the CALLERID(number)= I am still not getting the callerid aross voicepulse ... :-( |
| 04:11.43 | jeev | fender, is zaptel required to run asterisk? i tried disabling it in menuselect |
| 04:12.56 | HumanCell | mchou: using exten=s,1,NoOp(RINGGROUP) |
| 04:12.56 | HumanCell | exten=s,2,Set(CALLERID(number)=5556080269) |
| 04:12.56 | HumanCell | exten=s,n,Dial(SIP/5000&IAX2/5000&SIP/151112509@trunk_3,20) |
| 04:12.56 | HumanCell | Still doesn't work ... |
| 04:16.43 | [TK]D-Fender | jeev: No |
| 04:19.15 | jeev | ok |
| 04:24.27 | Johnakabean | jeev do "chkconfig zaptel off" |
| 04:24.32 | Johnakabean | in shell |
| 04:27.31 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 04:27.53 | [TK]D-Fender | JunK-Y: No clue... I suck at linux :) |
| 04:28.44 | mosty | junk-y: what kind of resources? |
| 04:31.17 | Johnakabean | Å°Ãé æìÃŮüûÐÐÐÐÃÃÊì ããéêæÄÃбÐÃøøÅÅ ÅÅÅÅÐü |
| 04:31.26 | Johnakabean | anyone alive? |
| 04:36.31 | *** join/#asterisk jameswf-home (n=james@cl-24.phx-01.us.sixxs.net) |
| 04:38.39 | *** join/#asterisk s0lid (n=s0lid@60.53.91.232) |
| 04:39.41 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 04:45.53 | *** part/#asterisk marklynch (n=markl@226.10.220.203.static.comindico.com.au) |
| 04:48.12 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 04:50.01 | CoffeeIV | I'm helping someone who apparently has this bug: http://bugs.digium.com/view.php?id=8565 and worked around it by adding "insecure=port,invite" to their conf files. What security implications might that have ? |
| 04:54.25 | kd8ikt | why are you using sip for asterisk to asterisk |
| 04:54.38 | kd8ikt | thats why there is such a thing call IAX |
| 04:55.12 | kd8ikt | called* |
| 05:02.44 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 05:08.44 | *** join/#asterisk Levonk (n=lk@adsl-75-62-131-235.dsl.lsan03.sbcglobal.net) |
| 05:19.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:22.16 | JunK-Y | mosty: that a kernel module is currently taking. |
| 05:23.04 | mosty | taking what, exactly? |
| 05:23.53 | *** part/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 05:27.22 | CoffeeIV | kd8bit: the provider uses SIP, I prefer IAX2 myself |
| 05:31.24 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 05:34.01 | *** join/#asterisk MrNaz (n=naz@122.110.231.224) |
| 05:36.12 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 05:42.03 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 05:45.33 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 05:45.51 | *** join/#asterisk ElSonico (n=tav@62.237.32.242) |
| 05:46.00 | k-man | what do you call a box that converts voip to normal phone? |
| 05:52.26 | drmessano | I don't know this joke |
| 05:52.30 | drmessano | :( |
| 05:53.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:53.28 | drmessano | Knock Knock |
| 05:53.58 | k-man | ATA? |
| 05:54.02 | k-man | i found it i think |
| 05:54.03 | drmessano | Knock Knock |
| 05:54.08 | k-man | whos there? |
| 05:54.11 | drmessano | Asterisk |
| 05:54.34 | k-man | asterisk who? |
| 05:54.41 | drmessano | Fix your NAT issues man, you have one-way audio |
| 06:01.25 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 06:05.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:05.52 | *** join/#asterisk s0lid (n=s0lid@60.53.91.232) |
| 06:08.35 | *** join/#asterisk Shotygun (n=thorn@84-16-228-45.internetserviceteam.com) |
| 06:09.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:12.37 | *** join/#asterisk Chicago (n=Chicago@c-98-223-62-171.hsd1.in.comcast.net) |
| 06:12.49 | Chicago | Hello |
| 06:15.43 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 06:23.01 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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| 06:39.32 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 06:45.19 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 06:56.16 | *** join/#asterisk Levonk (n=lk@adsl-75-62-128-223.dsl.lsan03.sbcglobal.net) |
| 06:58.53 | Madkiss | I actually want people to be able to call foobar@sip.mydomain.com; what do I do to create that SIP-URL? |
| 07:04.19 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:22.37 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 07:22.56 | MrTelephone | asterisk 1.2 is dead now? |
| 07:24.15 | mosty | it's old, not dead |
| 07:25.00 | MrTelephone | does 1.4 handle remote dtmf better? |
| 07:25.08 | MrTelephone | it sounds like it gets squelched in 1.2 |
| 07:25.59 | *** join/#asterisk zydoon (n=zydoon@41.225.143.63) |
| 07:26.12 | *** part/#asterisk zydoon (n=zydoon@41.225.143.63) |
| 07:28.05 | mosty | i assume you're talking about SIP- which method of dtmf are you using? |
| 07:28.56 | MrTelephone | it happens when someone on the pstn side pushes a digit |
| 07:29.08 | MrTelephone | the sangoma card or asterisk squelches it |
| 07:29.42 | MrTelephone | is there a list of features for 1.4 besides the changelog? |
| 07:29.55 | mosty | the changelog is a good start |
| 07:30.12 | MrTelephone | i didn't really want to know the code changes |
| 07:30.16 | MrTelephone | :-/ |
| 07:30.28 | MrTelephone | just crap like IMAP voicemail and whatever |
| 07:30.46 | kaldemar | MrTelephone: look at CHANGES and UPGRADE-1.2 in the source package |
| 07:31.12 | MrTelephone | ok |
| 07:31.34 | kaldemar | mm.. that would be UPGRADE.txt. UPGRADE-1.2.txt lists changes between 1.0 and 1.2. |
| 07:32.10 | MrTelephone | if the sip and zap are improved im a go for upgrade |
| 07:32.16 | MrTelephone | never know until you try though eh |
| 07:33.07 | MrTelephone | i opened it in notepad and there wasn't any <CR> yikes |
| 07:34.00 | kaldemar | well leave notepad alone. |
| 07:39.20 | *** join/#asterisk Levonk (n=lk@75.62.132.31) |
| 07:41.51 | *** join/#asterisk ToTo (n=ToTo@207.176.6.199) |
| 07:42.07 | MrTelephone | i kept my dialplans real simple so it shouldn't be too bad to switch |
| 07:43.17 | *** join/#asterisk laz_ (i=ken@black.hole-in-the.net) |
| 07:44.12 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 07:44.46 | raasdnil | hey all, what would be the best bet on a card to just give me timing. ie, the cheapest card I could buy that will give a good hardware zaptel timimg? |
| 07:45.39 | mosty | x100p? |
| 07:46.05 | Chicago | raasdnil: I am testing using dtdummy and an ethernet card... the packet carriers seem to be a less expensive way to get started. |
| 07:46.29 | florz | raasdnil: your question doesn't make sense |
| 07:46.36 | florz | what is "good zaptel timing"? |
| 07:47.19 | raasdnil | florz: sorry. Take out the good. Just hardware timing. |
| 07:47.45 | mosty | raasdnil, a recent linux kernel with hrtimers enabled and a recent version of zaptel 1.4 will work well with ztdummy |
| 07:47.53 | raasdnil | I have a few asterisk boxes, using ztdummy. I want to use hardware timing instead. I could buy some TDM400's or whatever, just wondering if there is a cheaper solution |
| 07:48.37 | mosty | a clone x100p from ebay would be cheap |
| 07:48.57 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 07:49.21 | *** join/#asterisk bazilek (n=bazil@mail.generation-p.com) |
| 07:50.01 | Chicago | raasdnil: Some of the other reading I have done says to get at least a EM64T Intel chip (IE Pentium 4 Extreme, or Core 2 Duo) so that you can use the HPET in the kernel for timing. |
| 07:50.32 | raasdnil | Chicago: well, I was having some issues, that's why i want to use hardware timing |
| 07:50.45 | tzafrir_laptop | Chicago, HPET does not give you timing on its own |
| 07:50.46 | Chicago | raasdnil: Describe the issues. |
| 07:51.02 | Chicago | tzafrir_laptop: Right |
| 07:51.05 | tzafrir_laptop | HPET is an echo canceller for existing Zaptel/DAHDI channels |
| 07:51.18 | Chicago | tzafrir_laptop: Just that HPET isn't available unless you have EM64T. |
| 07:51.28 | tzafrir_laptop | Chicago, huh? |
| 07:51.30 | florz | raasdnil: Well, of course hardware timing - how would you create a time scale in software? |
| 07:52.04 | raasdnil | Chicago: doing call center auto dialing with vicidial through a firewall to a pair of external DMZ'd asterisk boxes on IAX2 trunks that then go out through a NAT firewall to place calls for about 60 call center seats. Sometimes the calls don't dial. |
| 07:52.16 | raasdnil | sometimes they are all ok. |
| 07:52.30 | Chicago | tzafrir_laptop: The 2.6 kernel options for HPET don't affect all platforms... like plain pentium 4s don't take advantage of HPET... You have to get the later model Pentium4s with the 64bit memory or the more recent core 2 duo's etc... to use hpet. |
| 07:52.35 | raasdnil | gone through a few iterations of solutions, someone suggested maybe hardware timing could improve. |
| 07:53.07 | Chicago | raasdnil: What does tcpdump show? |
| 07:53.33 | Chicago | raasdnil: Or is the signal getting lost after you send the message to dial? |
| 07:53.45 | raasdnil | Chicago: shows the traffic flowing fine. the setup works, most of the time. But occasionally we'll get vicidial trying to dial 20 people at once to get a call. |
| 07:53.52 | florz | raasdnil: unless that someone has some detailed understanding of the inner workings of asterisk and zaptel and does indeed know about some strange interaction that could have such a result ... unless that's the case, just forget what that someone has said |
| 07:54.08 | raasdnil | I can run 40-60 concurrent calls through it. it is the vicidial autodialer that is not quite working |
| 07:54.30 | tzafrir_laptop | Chicago, err... not AFAIK. But I never used it anyway |
| 07:54.45 | tzafrir_laptop | (Use OSLEC if you don't want to be bitted by licensing pains) |
| 07:55.14 | raasdnil | florz: fair enough. That someone had a similar problem in another call center with the same situation and the problem went away after putting hardware timing into the boxes instead of ztdummy. Maybe co-incidence, that is why I am looking for the 'best' option on getting 4 cards that supply hardware timing |
| 07:55.59 | florz | but using a crystal on a PCB that has a digium logo on it won't make for any "better" timing than using a crystal on a PCB that has a <insert mainboard manufacturer here> logo on it |
| 07:56.00 | Chicago | raasdnil: Do you run a tickless kernel and what polling frequency do you set your kernels to and do you use kernel preemption? |
| 07:56.28 | raasdnil | florz: that makes sense |
| 07:56.42 | yang | tzafrir_laptop: How do those HFC cards provide timing ? |
| 07:57.05 | florz | raasdnil: there is no such thing as "hardware timing" ... or rather, there is nothing else than "hardware timing" - software only "runs" because hardware creates a clock ... |
| 07:57.10 | raasdnil | Chicago: I don't know the answers to those questions. I'll go and do some more study, that looks like some areas I could find out more on |
| 07:57.18 | florz | yang: the single port ones you mean? |
| 07:57.20 | raasdnil | florz: yeah |
| 07:57.33 | tzafrir_laptop | yang, err... I think florz can answer that better |
| 07:57.37 | florz | *g* |
| 07:58.17 | tzafrir_laptop | I'm not really sure how timing works on the HFC cards |
| 07:58.24 | raasdnil | Chicago: do you have some sites / docs / hints on where to study up on that? I am running clean installs (no gui) of Centos 4.6 on all of them with the latest versions of Asterisk 1.4 |
| 07:58.29 | tzafrir_laptop | (with our device the timing is a bit different) |
| 07:58.51 | Chicago | raasdnil: The help from make menuconfig gives an excellent description. |
| 07:59.00 | yang | florz: yep, single ported |
| 07:59.37 | raasdnil | Chicago: ok, I'll go check it out. thanks. |
| 07:59.44 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:00.03 | Chicago | raasdnil: Have you seen them in there before? |
| 08:00.11 | *** join/#asterisk telenieko (i=marc@telenieko.com) |
| 08:01.29 | telenieko | Hi ppl. I upgraded to asterisk 1.4.21 and now when I do ChanIsAvail() if the Channel is defined in a realtime table (i.e. sippeers) asterisk always says it's available thought it isn't (I share the table between servers). I could not find how to get the old behaviour, any ideas? |
| 08:02.59 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 08:03.26 | mosty | telenieko, i think you need qualify=yes, and i'm not sure if that works with realtime peers |
| 08:03.48 | telenieko | mosty: I'll try it. thanks |
| 08:07.53 | *** join/#asterisk qp (n=mg@82-71-8-45.dsl.in-addr.zen.co.uk) |
| 08:11.12 | qp | morning all |
| 08:15.56 | florz | yang: erm, sorry, was on the phone ... |
| 08:16.37 | florz | yang: well, the zaphfc driver uses an interrupt that's clocked by the isdn frame clock (8 kHz nominal) |
| 08:17.28 | tzafrir_laptop | the thing is that you upgrade to asterisk 1.4.21 and suddenly asterisk fails to start because zaphfc does not provide timing |
| 08:17.39 | tzafrir_laptop | (when the line is down, I guess) |
| 08:18.48 | florz | yang: which again is divided from some 12.something MHz (IIRC) crystal on the card, optionally locked to either an external clock source on the PCM bus (in NT mode) or to the frame clock on the S0 interface (in TE mode) |
| 08:19.05 | *** join/#asterisk ToTo (n=ToTo@207.176.6.192) |
| 08:19.10 | florz | well, actually the locking to the S0 frame clock is not optional in TE mode |
| 08:19.25 | florz | just that locking to an external PCM bus |
| 08:19.48 | Chicago | I asked for help in #asterisk-gui with setting up a custom trunk to Teliax. |
| 08:19.53 | Chicago | They are low traffic right now. |
| 08:20.37 | Chicago | I have a new setup, with Asterisk 1.4 and asterisk-gui 2.0SVN and need a little nudge in the right direction. |
| 08:21.00 | qp | anyone recommend the best solution for having several * boxes load balancing between themselves? also to handle things if one box goes "pop"? |
| 08:21.17 | qp | Open SER? |
| 08:21.22 | Chicago | System status shows that my Teliax trunk is registered with type iax to the proper host. |
| 08:21.47 | Chicago | But I am not even seeing the incoming calls show up with errors on the asterisk console. |
| 08:22.13 | Chicago | Is someone here using asterisk-gui 2.0 and Teliax whom can help me? |
| 08:27.01 | florz | tzafrir_laptop: hu? how that? using the same drivers, just different asterisk? |
| 08:27.43 | tzafrir_laptop | Asterisk will now fail at startup if it does not have a working timing source |
| 08:28.24 | florz | hmm - any clue what asterisk considers a "working timing source"? |
| 08:29.14 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:29.40 | florz | as long as ztcfg has been executed, I don't see how zaptel/asterisk could ever change the state as to whether it considers some zaphfc card "working" or "non-working" |
| 08:30.09 | tzafrir_laptop | http://svn.digium.com/view/asterisk?view=rev&revision=112689 |
| 08:31.49 | *** join/#asterisk sdr_ (n=sdr@mail.gsmreview.com) |
| 08:35.17 | sdr_ | hi, got problem with agi aplication dialing out. the problem is that if the chanell is closed i cant get to ANSWEREDTIME variable because when exec_dial returns the chanell is already closed and all variables are lost is there a work around or something? |
| 08:35.35 | *** join/#asterisk bad_duck_ (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 08:36.18 | tzafrir_laptop | florz, ==^ . It means that the zaptel clock does not "tick" |
| 08:36.43 | tzafrir_laptop | And thus some things in Asterisk that rely on it don't really work |
| 08:37.55 | florz | tzafrir_laptop: erm, so, asterisk doesn't complain but simply gets stuck? |
| 08:38.42 | tzafrir_laptop | before or after r112689 ? |
| 08:39.24 | tzafrir_laptop | After? fails to load. Before: music-on-hold and such won't work |
| 08:39.57 | *** join/#asterisk Levonk (n=lk@adsl-75-62-135-45.dsl.lsan03.sbcglobal.net) |
| 08:40.29 | florz | but then it fails to load, no matter what kind of lines are connected to the card, or how the card is configured!? |
| 08:40.38 | florz | erm, ?! rather |
| 08:44.23 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 08:45.37 | telenieko | Setting qualify=yes didn't fix the issue. asterisk says that anything defined in sippeers is "ok" on ChanIsAvail() ;\ |
| 08:45.57 | tzafrir_laptop | it fails to load if the card happens to be not connected anywhere |
| 08:46.25 | tzafrir_laptop | If you define that providing timing is part of a zaptel driver's job, then this is a bug of the driver |
| 08:47.34 | *** join/#asterisk Segnale007 (n=Segnale0@host187-248-dynamic.33-79-r.retail.telecomitalia.it) |
| 08:48.10 | florz | tzafrir_laptop: well, I'm just wondering how the timing "stops" |
| 08:49.10 | tzafrir_laptop | From what I understand: if the span is defined to use the provider's timing and the provider does not provide timing, there's no timing |
| 08:49.12 | florz | tzafrir_laptop: as in: how does asterisk/zaptel determine that there is no timing? |
| 08:49.22 | tzafrir_laptop | (at least with zaphfc) |
| 08:49.51 | florz | so, NT mode is not affected by this? |
| 08:50.02 | tzafrir_laptop | No, from my understanding |
| 08:51.07 | florz | but ... the interrupt counters do keep incrementing at ~ 8 kHz? |
| 08:51.30 | Chicago | Are there freely available custom voice menu prompts for download? |
| 08:51.34 | tzafrir_laptop | I'm not really sure. yang has such a card. Not me |
| 08:53.22 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:56.39 | qp | can anyone advise if the current vmware is OK to run * on for a pure voip solution? |
| 08:58.08 | mvanbaak | what vmware ? |
| 08:58.25 | mvanbaak | esx, server, esxi, virtualinfrastruct ? |
| 08:58.39 | qp | at the moment, server |
| 08:58.49 | mvanbaak | that works fine |
| 08:58.57 | qp | just speccing out * redundancy options |
| 08:59.12 | qp | virtual machines is so much easier |
| 08:59.25 | qp | what about the "timing issues" I read alot about, altho most articles are 2006 |
| 08:59.35 | mvanbaak | qp: you already have vmware server running? or are you planning a new setup ? |
| 08:59.44 | *** join/#asterisk luxxx77 (n=luxxx77@e176230221.adsl.alicedsl.de) |
| 08:59.58 | qp | already have it, seems ok, no more than 3 concurrent calls, but want to grow to much more |
| 09:00.03 | mvanbaak | ah ok |
| 09:00.10 | mvanbaak | well, it works fine in production here :) |
| 09:00.16 | *** join/#asterisk Oy90 (n=ivan@213.187.111.94) |
| 09:00.30 | qp | nice. do you use any clever load balancing or automatic switch overs for failing machines? |
| 09:00.35 | mvanbaak | we have 6 boxen with vmware server on it, and 3 of those boxen have asterisk virtual machines |
| 09:00.40 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:00.53 | qp | thats good to know. busy system? |
| 09:01.02 | mvanbaak | qp: we use dundi and an openbsd loadbalancer to do that |
| 09:01.15 | mvanbaak | not really busy, couple of thousand calls a day |
| 09:01.28 | qp | thats not "un busy" :) |
| 09:01.35 | mvanbaak | roughly 1500 sip registrations |
| 09:01.44 | qp | ok thanks, I will stick with vmware server until I have a reason not to use it any more :) |
| 09:02.30 | qp | lots of voicemail etc? |
| 09:03.04 | tzafrir_laptop | pings yang again to pick up the conversation |
| 09:03.54 | yang | yes |
| 09:04.13 | hi365 | looks for something simpler than sendmail |
| 09:04.45 | mvanbaak | qp: nah, normal voicemail volume for that many users. |
| 09:04.49 | mvanbaak | qp: no idea actually |
| 09:05.15 | qp | are you voip only? |
| 09:05.31 | mvanbaak | yup |
| 09:05.57 | yang | tzafrir_laptop: after recomilation music on hold works |
| 09:06.12 | yang | with the current setup |
| 09:06.34 | tzafrir_laptop | but at that time does zttest hang? |
| 09:07.20 | tzafrir_laptop | hi365, I generally prefer postfix |
| 09:08.00 | yang | I am wondering when does asterisk rotate log files defined in logger.conf ? |
| 09:08.13 | hi365 | tzafrir_laptop: that seems to be what people ware suggesting. can i use gmail as my smtp server? |
| 09:08.20 | tzafrir_laptop | "logger rotate"? |
| 09:09.44 | tzafrir_laptop | hi365, basically, yes . Though you have to pretend mail comes from you: |
| 09:10.03 | tzafrir_laptop | that is: use the "submit" service |
| 09:11.15 | *** join/#asterisk AutumnLeaves (n=AutumnLe@123.127.252.82) |
| 09:13.22 | tzafrir_laptop | yang, but this does not handle CSV CDR files |
| 09:14.59 | *** join/#asterisk angryuser (n=aster@88.140.126.251) |
| 09:15.05 | angryuser | good day |
| 09:15.22 | yang | tzafrir_laptop: it measures timing now |
| 09:15.23 | yang | --- Results after 50 passes --- |
| 09:15.24 | yang | Best: 99.998 -- Worst: 99.993 -- Average: 99.996345, Difference: 99.996345 |
| 09:15.30 | gr0mit | angryuser, are you sure? |
| 09:17.18 | angryuser | gr0mit, dont make me angry ;) |
| 09:21.07 | jblack | You wouldn't like him when he's angry |
| 09:27.37 | gr0mit | hehe! |
| 09:40.30 | *** join/#asterisk ahgindia (n=root@122.169.14.158) |
| 09:41.38 | *** join/#asterisk maxhbp2005 (n=maxhbp20@122.169.14.158) |
| 09:41.48 | ahgindia | hi |
| 09:41.55 | maxhbp2005 | hi |
| 09:42.03 | ahgindia | any help needed? |
| 09:42.12 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 09:42.17 | maxhbp2005 | i need some help about blind transfer |
| 09:42.18 | maxhbp2005 | ? |
| 09:42.29 | maxhbp2005 | is there anybody there? |
| 09:42.54 | ahgindia | i am there |
| 09:42.58 | ahgindia | te;; ,e |
| 09:43.02 | ahgindia | tell me |
| 09:43.35 | maxhbp2005 | i have agi script for inbound and outbound |
| 09:44.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 09:45.01 | maxhbp2005 | when i am calling a pstn call then my user connected to that number and then i am pressing *# and it asks transfer |
| 09:45.12 | maxhbp2005 | then i am entering transfer pstn number |
| 09:45.15 | maxhbp2005 | and it is working |
| 09:45.25 | maxhbp2005 | but in inbound that's the problem |
| 09:45.57 | maxhbp2005 | when i have received call and i am transfer that call then it is asks transfer but nothing happen |
| 09:46.08 | maxhbp2005 | can any body knows what's the issue is? |
| 09:46.42 | jblack | I don't understand you. |
| 09:46.58 | maxhbp2005 | let me explain again |
| 09:47.12 | maxhbp2005 | i have a script for outbound and inbound calling |
| 09:47.26 | maxhbp2005 | and it is working fine |
| 09:47.35 | maxhbp2005 | i have set *# for blind transfer |
| 09:47.50 | maxhbp2005 | blind transfer is working for outbound only |
| 09:48.06 | maxhbp2005 | it is not working while incoming call |
| 09:48.24 | maxhbp2005 | hi jblack |
| 09:48.29 | maxhbp2005 | can you please reply me |
| 09:48.31 | maxhbp2005 | ? |
| 09:48.34 | *** join/#asterisk Rico29 (n=Rico@ARennes-257-1-20-11.w81-53.abo.wanadoo.fr) |
| 09:49.00 | jblack | Ok. Adjust your Dial() setting for incoming if you're using blind transfer. |
| 09:49.24 | maxhbp2005 | i have passed Tt options in dial |
| 09:49.46 | jblack | In both dials? |
| 09:49.52 | maxhbp2005 | it is dialing from script after checking the incoming did and finds the sip user and the dial |
| 09:49.54 | jblack | (There's dialing out, and dial to dial to your sip lines) |
| 09:50.26 | maxhbp2005 | i have set tT in incoming call also |
| 09:51.02 | maxhbp2005 | it is playing prompt transfer in incoming but it not works |
| 09:51.06 | jblack | no ideas then. |
| 09:51.27 | maxhbp2005 | i have also set the __TRANSFER_CONTEXT=outgoing in script while the incoming call comes |
| 09:51.41 | angryuser | maxhbp2005, pastebin your extensions.conf and you features.conf |
| 09:52.00 | maxhbp2005 | ok |
| 09:52.06 | C4away | how does one make a patch file for a large number of files? |
| 09:52.59 | C4away | for example say I have /usr/src/foo/... and /usr/src/foo.modified/... each with many folders and files within |
| 09:53.16 | maxhbp2005 | [outgoing]exten=> _X.,1,Set(DYNAMIC_FEATURES=threeway) |
| 09:53.17 | maxhbp2005 | exten=> _X.,n,DeadAgi(asterisk_billing/calls.php) |
| 09:53.51 | maxhbp2005 | and same for incoming |
| 09:54.01 | maxhbp2005 | just context is different |
| 09:54.25 | *** part/#asterisk telenieko (i=marc@telenieko.com) |
| 09:54.27 | maxhbp2005 | features.conf [featuremap] |
| 09:54.28 | maxhbp2005 | blindxfer => *# |
| 09:55.12 | maxhbp2005 | hi angryuser |
| 09:55.32 | maxhbp2005 | can you please explain me |
| 09:56.25 | angryuser | maxhbp2005, please cli output during falied transfer |
| 09:56.57 | maxhbp2005 | it is just giving me the playback of transfer nothing else |
| 09:57.03 | maxhbp2005 | and else is the script |
| 09:58.21 | maxhbp2005 | == Setting global variable '1158' to '1158' |
| 09:58.23 | maxhbp2005 | <PROTECTED> |
| 09:58.24 | maxhbp2005 | <PROTECTED> |
| 09:58.26 | maxhbp2005 | <PROTECTED> |
| 09:58.28 | maxhbp2005 | <PROTECTED> |
| 09:58.29 | maxhbp2005 | <PROTECTED> |
| 09:58.31 | maxhbp2005 | <PROTECTED> |
| 09:58.32 | maxhbp2005 | <PROTECTED> |
| 09:58.34 | maxhbp2005 | <PROTECTED> |
| 09:58.40 | angryuser | ~pastebin |
| 09:58.41 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:00.11 | *** join/#asterisk buzzyd (n=psp-man@86.54.239.113) |
| 10:00.20 | *** join/#asterisk red1 (n=red1@158.65.49.60.klj04-home.tm.net.my) |
| 10:01.17 | maxhbp2005 | i have no idea about the pastebin |
| 10:02.07 | angryuser | maxhbp2005, do you have the option for the sip peer dtmfmode=rfc2833 ? also what kind of phone do you use ? |
| 10:02.23 | angryuser | maxhbp2005, now you have the idea |
| 10:02.57 | DarKnesS_WolF | ~pb |
| 10:02.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:03.12 | maxhbp2005 | yes i have realtime sip peers and set the dtmf mode=auto and my device linksys has also dtmf=auto |
| 10:03.59 | angryuser | try setting it to rfc2833 |
| 10:04.14 | maxhbp2005 | ok, just 1 min |
| 10:04.18 | maxhbp2005 | i am checking it |
| 10:05.31 | *** join/#asterisk oilinki (n=oil@ppp-124-120-23-222.revip2.asianet.co.th) |
| 10:07.57 | maxhbp2005 | it is not working |
| 10:08.12 | maxhbp2005 | http://paste.debian.net/15602/ |
| 10:09.32 | DarKnesS_WolF | maxhbp2005: what are u trying to do transfer |
| 10:09.33 | *** join/#asterisk Levonk (n=lk@75.62.134.95) |
| 10:10.19 | maxhbp2005 | when i have an incoming call i have pass the pstn number for testing after playback of transfer |
| 10:11.15 | maxhbp2005 | dtmf is also poperly set but the channel is not transfered to that number and my call is hangup |
| 10:13.24 | DarKnesS_WolF | maxhbp2005: show ur dialplan |
| 10:13.28 | DarKnesS_WolF | in pastbin |
| 10:13.32 | maxhbp2005 | yes |
| 10:14.16 | DarKnesS_WolF | maxhbp2005: ur using elaxtix / freepbx / trixbox ? |
| 10:14.36 | maxhbp2005 | i have asterisk 1.4.19 |
| 10:14.42 | maxhbp2005 | i haven't any pbx |
| 10:16.23 | maxhbp2005 | http://paste.debian.net/15603/ |
| 10:17.03 | maxhbp2005 | i have script that will dial |
| 10:19.51 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-352df038499bf8f6) |
| 10:22.38 | ahgindia | got any solution for transfer? |
| 10:22.44 | ahgindia | helooooooooooooooo |
| 10:22.47 | maxhbp2005 | no |
| 10:23.09 | maxhbp2005 | i am confused about that it is working in outbound but not in inbound |
| 10:23.20 | maxhbp2005 | both dial string are also proper |
| 10:24.03 | maxhbp2005 | is anybody having any idea about this? |
| 10:24.05 | ahgindia | so you can abort the transfer in inbound call |
| 10:24.20 | ahgindia | abort its use in inbound call.......... |
| 10:24.21 | ahgindia | :) |
| 10:24.23 | ahgindia | simple |
| 10:24.26 | ahgindia | isn't it |
| 10:24.39 | maxhbp2005 | but i need the solution of that |
| 10:24.51 | maxhbp2005 | i want to check what happening on asterisk |
| 10:25.34 | ahgindia | ok so u can try feviquick as a solution |
| 10:25.43 | maxhbp2005 | are you kidding |
| 10:25.45 | maxhbp2005 | ? |
| 10:25.46 | ahgindia | no |
| 10:25.49 | *** part/#asterisk Oy90 (n=ivan@213.187.111.94) |
| 10:25.49 | ahgindia | its possible |
| 10:25.51 | ahgindia | in asterisk |
| 10:25.58 | ahgindia | just try that |
| 10:27.02 | maxhbp2005 | thank you very much everybody, |
| 10:27.07 | *** part/#asterisk maxhbp2005 (n=maxhbp20@122.169.14.158) |
| 10:28.30 | *** join/#asterisk maxhbp2005 (n=maxhbp20@122.169.14.158) |
| 10:29.55 | *** join/#asterisk ToTo (n=ToTo@207.176.6.199) |
| 10:30.08 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 10:36.17 | *** join/#asterisk Johnakabean (n=Johnakab@pool-72-82-111-106.nrflva.east.verizon.net) |
| 10:40.28 | *** join/#asterisk ToTo (n=ToTo@207.176.6.199) |
| 10:41.51 | Great_Anta_Baka | does 1.4 asterisk have a web interface that you can check out the cdr records? |
| 10:45.34 | kaldemar | no |
| 10:46.07 | Great_Anta_Baka | what things can you view from the 1.4 web interfaces? |
| 10:46.35 | Great_Anta_Baka | i know you can check the manager |
| 10:48.14 | *** join/#asterisk ToTo (n=ToTo@207.176.6.210) |
| 10:50.04 | *** join/#asterisk maxhbp2005 (n=maxhbp20@122.169.14.158) |
| 10:50.09 | *** join/#asterisk Levonk (n=lk@adsl-75-62-130-34.dsl.lsan03.sbcglobal.net) |
| 10:51.12 | tzafrir_laptop | ~ask |
| 10:51.12 | jbot | i heard ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:53.16 | tzafrir_laptop | Didn't take a clue, I guess |
| 10:59.18 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 10:59.24 | mandh | hello all |
| 10:59.53 | mandh | i have zap1-1, zap2-1 ,i have lot of abandoned calls |
| 11:00.38 | mandh | 40% of them on zap2-1 and 10% on zap1-1 |
| 11:01.17 | mandh | when debug calls i found that most of calls comming without caller-id |
| 11:01.42 | mandh | on the zap2 |
| 11:02.30 | tzafrir_laptop | soft hangup Zap/1-1 |
| 11:02.44 | Great_Anta_Baka | ~ask what things can you view from the 1.4 web interfaces? |
| 11:02.47 | tzafrir_laptop | are those analog trunks? |
| 11:02.56 | Great_Anta_Baka | mmm |
| 11:03.05 | Great_Anta_Baka | how do you use jbot? |
| 11:03.20 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 11:03.20 | tzafrir_laptop | jbot has no entry for "sk what things can you view from the 1.4 web interfaces?" |
| 11:03.56 | tzafrir_laptop | ask jbot something it does know |
| 11:04.43 | Great_Anta_Baka | eh |
| 11:04.59 | Great_Anta_Baka | can you give me an example |
| 11:05.07 | kaldemar | Great_Anta_Baka: http://www.asterisknow.org/image/tid/58 |
| 11:05.08 | Great_Anta_Baka | i dont know what jbot knows |
| 11:05.12 | tzafrir_laptop | ~Great_Anta_Baka |
| 11:05.13 | jbot | rumour has it, great_anta_baka is an old wooden ship. |
| 11:05.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 11:05.56 | Great_Anta_Baka | lol |
| 11:05.59 | Great_Anta_Baka | i see |
| 11:06.07 | tzafrir_laptop | ~jbot |
| 11:06.08 | jbot | i guess jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 11:06.20 | Great_Anta_Baka | thanks kaldemar |
| 11:08.57 | *** join/#asterisk Knightfal (n=Knightfa@75.142.144.45) |
| 11:09.12 | kaldemar | using a search engine doesn't hurt. |
| 11:10.03 | Great_Anta_Baka | indeed google is my friend |
| 11:15.58 | *** join/#asterisk sack (n=sack@101.Red-83-55-222.dynamicIP.rima-tde.net) |
| 11:32.02 | yang | I have a Siemens handset which has 6 phone extensions - they also listen on the same IP and SIP port, but is this proper - http://www.pastebin.sk/sk/7915/ |
| 11:32.08 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 11:37.10 | jblack | Usually a device that does multiple sip accounts puts each account on a different port. |
| 11:37.21 | DarKnesS_WolF | yang: yes i do have a snom m3 dect with 2 accounts and it the same output as urs and works fine. |
| 11:37.30 | yang | hmmm |
| 11:37.35 | DarKnesS_WolF | jblack: don't think so |
| 11:37.47 | jblack | Ok. You don't think so. No problem. |
| 11:38.10 | yang | I have the SIP setting for the PBX but this one is always 5060....and the Siemens handset base station is only bind to one ip |
| 11:38.31 | DarKnesS_WolF | http://pastebin.ca/1185206 |
| 11:39.01 | DarKnesS_WolF | yang: jblack i think the phone handeling which ext. should ring " for diffrunt rings " |
| 11:39.14 | *** join/#asterisk pitombera (n=pitomber@unaffiliated/pitombera) |
| 11:39.24 | yang | The problem is that I cannot investigate what makes the calls deaf sometimes....the other side doesn't hear the caller, but on the next call it works fine... |
| 11:39.37 | DarKnesS_WolF | yang: how ? i should have multi account page each one will have it is own IP for the realm |
| 11:40.00 | DarKnesS_WolF | s/i/it |
| 11:43.18 | yang | and such problems appear on all handsets |
| 11:43.40 | DarKnesS_WolF | yang: try to change the ports in each identiy or let hte phone use random one. |
| 11:43.57 | DarKnesS_WolF | inever had such problem with snom/ giptel with multi accounts on same set |
| 11:44.23 | yang | well snom is known to be good, but who knows how Siemens works |
| 11:44.34 | *** join/#asterisk JenniferAkemi (n=akemi@MTLXPQAK-1178074603.sdsl.bell.ca) |
| 11:45.27 | DarKnesS_WolF | yang: get snom them :P |
| 11:45.43 | yang | Yeah well, I am not ordering them |
| 11:45.57 | DarKnesS_WolF | yang: if u have one account |
| 11:46.00 | yang | Its my boss who gets the cheapest hardware |
| 11:46.01 | DarKnesS_WolF | everything wroks fine? |
| 11:46.13 | DarKnesS_WolF | siemens is cheaper than SNOM !? |
| 11:46.14 | DarKnesS_WolF | noway ! |
| 11:46.34 | yang | Does Snom produce wireless phones at all? |
| 11:47.04 | DarKnesS_WolF | yang: yes |
| 11:47.05 | DarKnesS_WolF | M3 |
| 11:47.07 | DarKnesS_WolF | Dect |
| 11:47.07 | angryuser | is it normal when macro exits it is going back to primary dialplan and continue execute it ? |
| 11:47.21 | *** part/#asterisk pitombera (n=pitomber@unaffiliated/pitombera) |
| 11:47.24 | angryuser | siemens is cheaper ans they are good |
| 11:47.39 | DarKnesS_WolF | one base supports up to 8 dect handset and also it has repeaters |
| 11:48.14 | angryuser | i got siemens dect and snoms (not dect) al working good |
| 11:49.20 | yang | maybe my settings are wrong - i have the same for all handsets - http://www.pastebin.sk/sk/7917/ |
| 11:49.51 | yang | The phone doesn't work if I don't define "proxy" there , however I don't use any real proxy |
| 11:50.37 | yang | DarKnesS_WolF: do you think that I could change "proxy server port" to different value for each handset? |
| 11:51.34 | DarKnesS_WolF | yang: read teh set manual |
| 11:54.30 | angryuser | is there any wayt to force macro to do not return to original context ? |
| 11:54.46 | *** join/#asterisk ToTo (n=ToTo@207.176.6.210) |
| 11:54.47 | angryuser | *after it is done it's job* |
| 11:54.49 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 11:55.17 | r0land | hello all |
| 11:55.35 | DarKnesS_WolF | angryuser: to hang up ? in the end of the macro ? |
| 11:55.36 | DarKnesS_WolF | hangup |
| 11:56.15 | r0land | can any1 help out with an authentication prob! |
| 11:56.26 | tzafrir_laptop | ~anyone |
| 11:56.27 | jbot | *** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do. |
| 11:56.59 | r0land | im tryig to do the following: i have 10 sip peers i want 8 of these peers to b able to call the 9th normaly.. bu the 10th to require a password if he wanna call the 9th |
| 11:57.07 | r0land | does tht make any senes to u ? |
| 11:57.34 | tzafrir_laptop | is still parsing |
| 11:57.51 | angryuser | well i got one macro for call out, all is fine and call completes (busy noanswer answered) after it jumps back to original context and continue it's execution ;) |
| 11:58.24 | angryuser | r0land, set a different context for that person |
| 11:58.55 | tzafrir_laptop | r0land, generally use something of the sort of Authenticate() to check for a password |
| 12:00.38 | angryuser | tzafrir_laptop, it will work for *all* peers, i think he need to create another context copy rules there and modify one with Authenticate |
| 12:01.12 | tzafrir_laptop | sure. A different context is needed anyway |
| 12:01.24 | angryuser | or set vars in astdb for each peers and verify before call |
| 12:05.18 | r0land | angryuser yes i thought abotu tht! but other users may not b able to call it ! |
| 12:05.33 | angryuser | r0land, sure they will |
| 12:05.47 | r0land | tzafrir_laptop yes i tried tht. but tht way every1 would have to enter tht passworfd! though i just want to deny access to one peer |
| 12:06.12 | angryuser | r0land, core show application Dial() under cli |
| 12:07.04 | r0land | angryuser Your application(s) is (are) not registered |
| 12:07.31 | angryuser | r0land, then Dial |
| 12:08.41 | *** join/#asterisk ElSonico (n=tav@a91-153-124-254.elisa-laajakaista.fi) |
| 12:08.54 | *** join/#asterisk matsk (n=Mats@host-90-235-32-186.mobileonline.telia.com) |
| 12:08.55 | r0land | angryuser im not following u :S |
| 12:09.11 | angryuser | r0land, just create another context copy all dial rules for each sip account there, set 'context' for the 2 peers |
| 12:09.17 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:09.25 | angryuser | r0land, to you new fresh context |
| 12:09.31 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 12:09.37 | r0land | angryuser ya but to do so! ill deny other sip thts registered in other context! to b able to call these 2.. |
| 12:09.41 | r0land | unless if i "include" |
| 12:09.48 | r0land | and if i included! its like i havent done anything useful |
| 12:10.13 | angryuser | r0land, and modify dialin rules for peer needed |
| 12:10.43 | angryuser | pastebin your extensions.conf and sip.conf please |
| 12:10.53 | r0land | angryuser ok just a sec |
| 12:12.49 | angryuser | you can allways use goto to speed up |
| 12:13.06 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 12:13.23 | r0land | angryuser http://pastebin.com/d4ea01866 |
| 12:13.35 | *** join/#asterisk ElSonico (n=tav@a91-153-124-254.elisa-laajakaista.fi) |
| 12:13.52 | r0land | i dont want extension 300 to b able to call "03,04,05,06" without entering a code |
| 12:14.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:14.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:14.34 | angryuser | r0land, ok |
| 12:15.07 | r0land | thts it |
| 12:16.24 | r0land | angryuser i tried adding "authenticate" in sipura-line! but tht made all other peers to HAVE to use the same password as the extension 300 to access those lines |
| 12:17.11 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:17.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:17.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:20.20 | r0land | angryuser any advice? |
| 12:20.49 | angryuser | r0land, yes sec |
| 12:20.59 | r0land | angryuser gr8 thanks :) |
| 12:22.21 | angryuser | r0land, http://pastebin.ca/1185241 create context [reistricted] |
| 12:22.21 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:23.11 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 12:23.12 | angryuser | and modify sip.conf as shown |
| 12:23.58 | r0land | angryuser i cant access pastebin.ca :( |
| 12:24.16 | *** join/#asterisk Dead_Assclown (n=KnoX@modemcable124.92-203-24.mc.videotron.ca) |
| 12:24.16 | r0land | is it possible if you use pastebin.com instead ?! |
| 12:25.32 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 12:25.56 | angryuser | r0land, http://pastebin.com/m384db73c |
| 12:26.22 | *** join/#asterisk Levonk (n=lk@75.62.132.108) |
| 12:26.51 | [TK]D-Fender | angryuser: Unnecessarily large.... that could have been shrunk a LOT. |
| 12:27.48 | angryuser | [TK]D-Fender, yes , but let's try to make it work firs :) |
| 12:28.18 | [TK]D-Fender | angryuser: Guess I won't mention the broken bits ;) |
| 12:29.19 | angryuser | [TK]D-Fender, hey tk by the way how to stop macro going back to original exten after execution, i cant find the option ;( or it is supposed to be like that ? |
| 12:29.38 | [TK]D-Fender | angryuser: "Hangup" :) |
| 12:30.26 | [TK]D-Fender | angryuser: and yes its supposed to return, thats why its a macro and not a "set X args and jump" |
| 12:33.37 | angryuser | r0land, so roland, done ? |
| 12:34.36 | r0land | angryuser having trouble with the other extnsions |
| 12:35.01 | r0land | one question! i added exten => 301,1,authenticate(11) <<-- do i change the priority ?! |
| 12:35.34 | lmadsen | first priority best practice is to make it a NoOp() or Verbose(), and start all real lines on the 2nd line, with the 'n' priority |
| 12:36.04 | [TK]D-Fender | r0land: with your KEYBOARD |
| 12:36.08 | angryuser | r0land, under cli 'core show authenticate' gives you all the syntax |
| 12:36.39 | [TK]D-Fender | r0land: exten => 301,1234,authenticate(11) <- OMG, its a different priority!!! |
| 12:36.55 | angryuser | core show application authenticate ^^;) |
| 12:36.59 | [TK]D-Fender | lmadsen: EW, useless NoOps! |
| 12:37.22 | lmadsen | Its not useless.... I hate using a real application at the first priority |
| 12:37.28 | lmadsen | use it for documentation |
| 12:37.34 | creativx | why do you not like 1st pri! |
| 12:37.48 | r0land | getting me ost here:) |
| 12:38.05 | SteveTotaro | hi roland |
| 12:38.06 | [TK]D-Fender | lmadsen: that what COMMENT lines are for. You're slowing down your dialplan execution, throwing out CLI junk, and ";" is FAR more efficient for making a comment :) |
| 12:38.13 | SteveTotaro | if you want real answers, ask me |
| 12:38.19 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:38.24 | *** join/#asterisk Levonk (n=lk@adsl-75-62-132-169.dsl.lsan03.sbcglobal.net) |
| 12:38.40 | [TK]D-Fender | SteveTotaro: 42 <- |
| 12:38.41 | r0land | SteveTotaro hello!! |
| 12:38.43 | [TK]D-Fender | huzzah! |
| 12:38.52 | r0land | SteveTotaro you helped me out once before! on asterisk's mailing list |
| 12:38.57 | *** part/#asterisk LemensTS (i=LemensTS@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
| 12:39.02 | SteveTotaro | yup, 42 but from the old days |
| 12:39.09 | r0land | SteveTotaro you guided me to the authenticate issue! |
| 12:39.17 | SteveTotaro | yeah, how could i forget you r0land |
| 12:40.01 | [TK]D-Fender | SteveTotaro: Yeah, you can't afford that kind of psycho-therapy! |
| 12:40.15 | r0land | SteveTotaro http://pastebin.com/d4ea01866 <<-- this is my current config! i want to deny access to 301 to b able to call "304,305,306,307,308" |
| 12:40.23 | SteveTotaro | i subscribe to freudian theory |
| 12:40.24 | r0land | SteveTotaro unless if the caller enters a password |
| 12:40.31 | *** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni) |
| 12:40.40 | r0land | angryuser was kind enough to give me an advice and a pastebin about it! but i just tried it .. didnt work :( |
| 12:40.55 | SteveTotaro | put 301 in a separate context with no include to those extensions |
| 12:41.21 | SteveTotaro | oh, a password |
| 12:41.27 | r0land | SteveTotaro yes a pass |
| 12:41.54 | SteveTotaro | separate context with authenticate() and then a goto the main context |
| 12:41.55 | r0land | SteveTotaro tried exten => 301,1,authenticate(11) but tht made all other peers to ask for tht pass not just 301 |
| 12:43.06 | r0land | SteveTotaro lost me thre! |
| 12:43.21 | [TK]D-Fender | r0land: put it in ANOTHER CONTEXT!! |
| 12:43.40 | SteveTotaro | set up exten 301 in sip.conf to have a separate context from the rest, spa if memory serves me correctly |
| 12:43.40 | r0land | [TK]D-Fender if i put it in another context it wont call other peers! |
| 12:43.45 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 12:43.48 | r0land | SteveTotaro i did tht! |
| 12:43.57 | SteveTotaro | ok but then use goto() |
| 12:43.59 | r0land | SteveTotaro i followed angryuser 's advice and set it to a "restricted" context |
| 12:44.02 | SteveTotaro | after authenticate |
| 12:44.05 | *** join/#asterisk mltlnx (n=mltlnx@96.232.20.139) |
| 12:44.05 | r0land | ah |
| 12:44.21 | r0land | hm tht makes sense |
| 12:44.37 | SteveTotaro | goto(context,exten,prio) i believe.... |
| 12:46.39 | r0land | SteveTotaro i add it right after authenticate in the new context |
| 12:46.46 | [TK]D-Fender | r0land: http://pastebin.com/mea4c393 |
| 12:46.51 | SteveTotaro | yeah, should be short and simple |
| 12:47.29 | [TK]D-Fender | So simple its SAD |
| 12:47.49 | SteveTotaro | but r0land has enthusiasm! |
| 12:48.01 | SteveTotaro | he has been bitten by the asterisk bug |
| 12:48.11 | r0land | sorry for the newbie questions |
| 12:48.25 | [TK]D-Fender | r0land: You've been using * for HOW long now? |
| 12:48.30 | r0land | SteveTotaro been working on this for a while now.. im trying to add every feature in asterisk to my current setup so i can get a better grop for it |
| 12:48.35 | SteveTotaro | well, i suggest you keep off irc for a year or two and stick to the wiki |
| 12:48.37 | r0land | [TK]D-Fender 5 weeks |
| 12:48.41 | [TK]D-Fender | r0land: If you don't understand the dialplan you're pretty much screwed |
| 12:48.54 | [TK]D-Fender | r0land: 5 weeks? No, I've seen you here FAR longer than that... |
| 12:49.00 | yang | What kind of error is this - Aug 26 13:58:56 lineamedia asterisk[2203]: rc_avpair_new: unknown attribute 1490026597 |
| 12:49.02 | r0land | [TK]D-Fender 14 str8 days.. or 5 weeks on and off |
| 12:49.21 | SteveTotaro | yang, that is a "funky" error |
| 12:50.24 | SteveTotaro | r0land has been on craig's list too much with the "str8" thing. (sure you are buddy:P) |
| 12:51.00 | yang | SteveTotaro: as google tells me its the Audio error - this is exactly the problem i have no audio |
| 12:51.21 | SteveTotaro | no audio ain't no good for asterisk |
| 12:51.23 | r0land | lol |
| 12:51.48 | [TK]D-Fender | SteveTotaro: Over negatived... |
| 12:52.24 | SteveTotaro | sorry, i live on the cusp of the hood |
| 12:53.09 | SteveTotaro | yang, what kind of devices are you using, i have never seen an error like that |
| 12:54.12 | yang | no audio only occasionally - the call is established and they cannot hear, while I can hear them, and if i redial the number again the sound usually works for both sides |
| 12:54.41 | yang | SteveTotaro: HFC cards and Siemens Gigaset wireless handsets |
| 12:54.52 | angryuser | yang, misdn ? |
| 12:54.58 | yang | no zaptel |
| 12:55.24 | SteveTotaro | yeah, i am not familiar with misdn whatsoever, it is totally greek to me |
| 12:55.28 | SteveTotaro | sorry |
| 12:56.08 | angryuser | yang, i had problems like that with siemens, check your udp ports, set them to different ones, also tcp |
| 12:56.17 | SteveTotaro | bristuff i have messed with too much, nobody in the US should have a bri imho |
| 12:56.40 | SteveTotaro | not in washington dc anyways |
| 12:56.43 | x86 | I know some people running BRI still in Illinois |
| 12:56.50 | angryuser | SteveTotaro, should i say the same thing for misdn :) |
| 12:56.52 | x86 | it's rare, but it does happen still |
| 12:57.37 | *** join/#asterisk maxhbp2005 (n=maxhbp20@122.169.14.158) |
| 12:57.44 | *** join/#asterisk eliel (n=eliel@200.61.172.61) |
| 12:58.15 | SteveTotaro | i have a client in vienna va (washington dc urban sprawl) with two bris |
| 12:58.30 | SteveTotaro | and then they wanted me to add four analog lines...... ahhhh |
| 12:58.52 | *** part/#asterisk maxhbp2005 (n=maxhbp20@122.169.14.158) |
| 12:58.55 | yang | angryuser: bad Siemens! |
| 12:59.16 | SteveTotaro | i stay aways from semens |
| 12:59.30 | SteveTotaro | r0land can help though |
| 13:00.06 | r0land | ? |
| 13:00.26 | SteveTotaro | bad joke, nm |
| 13:00.32 | r0land | :) |
| 13:00.56 | r0land | SteveTotaro the pastebin [TK]D-Fender gave! the way i got it! didnt do a thing! |
| 13:01.16 | r0land | since context "normalstuff" had the peers tht i DONT want ppl to b able to call |
| 13:01.35 | [TK]D-Fender | r0land: Hopefully you get the idea anyways. |
| 13:01.43 | r0land | which are 03,04,05,06 that in turn redirect to 304,305,306,307 |
| 13:01.49 | r0land | respectively |
| 13:02.36 | [TK]D-Fender | r0land: don't "redirect", use INCLUDES |
| 13:02.38 | r0land | the way i get it it only made ppl who call 301, to have to enter a password to access ANY other peer |
| 13:02.43 | [TK]D-Fender | Goto for this is jsut silly |
| 13:02.48 | yang | angryuser: UDP ports related to number assignment? |
| 13:03.11 | r0land | [TK]D-Fender lemme explain again whts the prob im facing |
| 13:03.54 | SteveTotaro | goto brings me back to basic ;) |
| 13:04.14 | [TK]D-Fender | r0land: Your problem is that you aren't splitting up your contexts in a smart way |
| 13:04.18 | angryuser | yang, be sure that udp ports are not out of range! and not set to !random! |
| 13:06.24 | *** join/#asterisk korihor (n=korihor@200.7.99.2) |
| 13:06.46 | r0land | [TK]D-Fender http://pastebin.com/m309bbedd |
| 13:06.59 | r0land | [TK]D-Fender yes thts one thing i tottally agree with! |
| 13:07.05 | r0land | [TK]D-Fender which is why im here.. |
| 13:07.19 | r0land | [TK]D-Fender could check this please http://pastebin.com/m309bbedd |
| 13:07.52 | SteveTotaro | contexts are the most important part of the dialplan |
| 13:07.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.02 | SteveTotaro | beyond that, it is just commands and syntax |
| 13:08.39 | [TK]D-Fender | r0land: [sipura-line] should NOT include [spa]. That is retarded |
| 13:08.57 | SteveTotaro | retarded offends me |
| 13:09.09 | r0land | [TK]D-Fender if t didnt include it! if the caller to 301, dialed lets say 100! how would it know where to go! |
| 13:09.45 | [TK]D-Fender | r0land: and your "301" exten priorities are screwed up BAD. |
| 13:10.05 | SteveTotaro | use n if you can't get your priorities right |
| 13:10.18 | *** join/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 13:10.21 | *** part/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 13:10.23 | yang | angryuser: does Listen port for VOIP connections RTP port must match the ports in rtp.conf (geez how could i misslook that) |
| 13:11.01 | yang | angryuser: thanks a lot ! |
| 13:11.17 | r0land | [TK]D-Fender <r0land> [TK]D-Fender if t didnt include it! if the caller to 301, dialed lets say 100! how would it know where to go! |
| 13:12.10 | [TK]D-Fender | r0land: You are so spun around its friggen ridiculous |
| 13:12.21 | SteveTotaro | i say get off irc |
| 13:12.30 | SteveTotaro | learn asterisk, just keep trying stuff |
| 13:12.46 | SteveTotaro | until you say "ahhh, i get it now" |
| 13:13.01 | SteveTotaro | you will be much better off in the long run |
| 13:13.18 | SteveTotaro | i have been doing asterisk for seven years but.... |
| 13:13.23 | SteveTotaro | ~stevetotaro |
| 13:13.24 | jbot | hmm... stevetotaro is an IRC nub |
| 13:13.33 | Katty | morning |
| 13:13.44 | SteveTotaro | bonjour |
| 13:13.46 | [TK]D-Fender | r0land: http://pastebin.com/m19587edc |
| 13:13.52 | [TK]D-Fender | r0land: Merry Christmas |
| 13:14.01 | Katty | oh? |
| 13:14.05 | Katty | i missed halloween and thanksgiving. |
| 13:14.06 | creativx | hello Katty |
| 13:14.07 | Katty | dangit. |
| 13:14.12 | [TK]D-Fender | Katty: Mew. |
| 13:14.19 | creativx | there will be others next year Katty. fear not |
| 13:14.22 | Katty | [TK]D-Fender: mew. |
| 13:14.28 | Katty | creativx: that's not the point :< |
| 13:14.38 | SteveTotaro | plus you missed the extra 10 pounds you would have put on |
| 13:14.46 | Katty | that's true ;) |
| 13:14.47 | SteveTotaro | there is always a silver lining |
| 13:14.57 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:14.59 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 13:15.18 | r0land | [TK]D-Fender thanks :) |
| 13:15.22 | Katty | [intra]lanman: mew. |
| 13:15.36 | Katty | [intra]lanman: how are the mini-lans. |
| 13:15.53 | [intra]lanman | Katty: the lanbrats are good... thanks for asking |
| 13:16.00 | Katty | [intra]lanman: good to hear. |
| 13:16.10 | Katty | [intra]lanman: the petting zoo doing okay too? |
| 13:17.05 | [intra]lanman | hah, yeah |
| 13:17.13 | Katty | cheers (= |
| 13:17.29 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:17.52 | Katty | should clean her office today due to company coming over. |
| 13:18.11 | SteveTotaro | get your minions to clean it |
| 13:18.30 | Katty | there's a thought. |
| 13:18.33 | Katty | i'll bring my ferrets in! |
| 13:18.49 | SteveTotaro | ferrets are cool |
| 13:18.53 | Maliuta | MMOD(tm) |
| 13:19.00 | SteveTotaro | ever since beast master |
| 13:19.01 | Katty | if by cool you mean completely adorable. |
| 13:19.04 | Maliuta | Minion Monkeys of Dooooooom(tm) |
| 13:19.12 | Maliuta | they run _my_ * system for me |
| 13:19.13 | Corydon76-dig | and stinky |
| 13:19.32 | Katty | Corydon76-dig: we keep ours clean. |
| 13:19.35 | SteveTotaro | yeah, i bet they do stink but they steal stuff, that is cool |
| 13:19.52 | SteveTotaro | they "ferret" things away |
| 13:20.03 | SteveTotaro | they have their own verb |
| 13:20.06 | Katty | and bounce. don't forget the bouncing. |
| 13:20.16 | Katty | random ankle attacking. |
| 13:20.23 | Katty | and the random falling asleep everywhere. |
| 13:20.23 | SteveTotaro | brooklyn bounce? |
| 13:20.39 | Katty | and their toothy lil yawns :> |
| 13:20.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 13:21.09 | SteveTotaro | my pit bull would eat them up |
| 13:21.26 | Katty | or snorrgle them to death. |
| 13:21.50 | SteveTotaro | he bats at little creatures when they move |
| 13:21.56 | Katty | oh )= |
| 13:22.03 | SteveTotaro | he thinks he is being gentle |
| 13:22.08 | Katty | eh. |
| 13:22.13 | Katty | i'm sure he does. |
| 13:22.27 | SteveTotaro | well an 85# pit bull has a different idea of gentle |
| 13:22.33 | Katty | of course ;) |
| 13:22.50 | Katty | i always thought pit bulls had a genetic defect of slight mental... issues. |
| 13:23.10 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:23.10 | SteveTotaro | that is generally due to ignorance, no offense |
| 13:23.25 | Katty | nods |
| 13:23.28 | SteveTotaro | many are so inbred though, just like any "pure bred" dog |
| 13:23.30 | Katty | i'm glad that got cleared up. |
| 13:24.01 | Katty | i think there's a lot of misconceptions about a great many dogs due to propaganda and tv. |
| 13:24.24 | SteveTotaro | well my brother is a cop in PG county MD where pit bulls are outlawed |
| 13:24.42 | Katty | ferrets are also outlawed in some areas. |
| 13:24.46 | [TK]D-Fender | Good : a cat. Better : a dog. Best : a dog that eats cats. :D |
| 13:24.55 | Katty | [TK]D-Fender: :< |
| 13:25.05 | Katty | [TK]D-Fender: can't we all just get along!? |
| 13:25.09 | Katty | [TK]D-Fender: -Imp |
| 13:25.10 | SteveTotaro | not really fender, my dog killed my ex gf's cat |
| 13:25.16 | r0land | [TK]D-Fender thanks for before.. but could u explain to me wht this counts for! just to understand! _[123]XX |
| 13:25.18 | SteveTotaro | hence the ex |
| 13:25.31 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:25.41 | SteveTotaro | prefix 123 then two more digits |
| 13:25.50 | r0land | i dunno wht does the [ ] stands for! i mean if i set _1XX <<-- i get this..! but whts with the [ ] |
| 13:25.58 | r0land | ah ok |
| 13:25.59 | SteveTotaro | prefix |
| 13:26.01 | r0land | i got it |
| 13:26.03 | r0land | thanks ) |
| 13:26.16 | SteveTotaro | you should really read r0land |
| 13:26.28 | r0land | SteveTotaro i know i have alot to catch up |
| 13:26.29 | SteveTotaro | not irc but the book or wiki or whatever |
| 13:26.32 | r0land | thanks for the help u guys :) |
| 13:26.41 | r0land | SteveTotaro could u guide me to the wiki plz! i dont have its link.. |
| 13:26.45 | r0land | i just visit asteriskguru |
| 13:26.51 | SteveTotaro | ~wiki |
| 13:27.07 | SteveTotaro | huh, www.voip-info.org |
| 13:27.15 | r0land | k |
| 13:27.17 | r0land | thanks :) |
| 13:27.17 | [TK]D-Fender | Katty: I believe in animal rights : http://www.welaf.com/funny-picture-12118.html |
| 13:27.32 | r0land | thanks [TK]D-Fender SteveTotaro :) appreciate the help |
| 13:27.40 | [TK]D-Fender | SteveTotaro>prefix 123 then two more digits <- um, NO |
| 13:27.42 | Katty | tkbeat: :< |
| 13:27.44 | Katty | oh. |
| 13:27.49 | [TK]D-Fender | r0land: 1xx + 2xx + 3xx all in one |
| 13:27.49 | Katty | [TK]D-Fender: :< |
| 13:27.55 | Katty | [TK]D-Fender: did you find that off reddit |
| 13:28.05 | [TK]D-Fender | Katty: GOOGLE :) |
| 13:28.06 | r0land | to b honest [TK]D-Fender s pastebin didnt work! incoming calls to 301 kept on ringing without answering which is weird.. but i understood wht he did and ill edit mine accordingly |
| 13:28.11 | [TK]D-Fender | ~[TK]D-Fender |
| 13:28.11 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 13:28.29 | SteveTotaro | i mean match on that prefix not prepend |
| 13:28.45 | [TK]D-Fender | r0land: May need a TINY tweak becase your description is so busted, but it'd better not grow by more that 3 lines. |
| 13:28.55 | [TK]D-Fender | SteveTotaro: {123] is not a prefix. |
| 13:29.04 | [TK]D-Fender | SteveTotaro: it is a digit LIST. |
| 13:29.11 | SteveTotaro | what is {123] i think that is wrong |
| 13:29.31 | [TK]D-Fender | SteveTotaro: [123]XX = 1XX or 2XX or 3XX |
| 13:29.31 | SteveTotaro | i think that would fail {123] |
| 13:29.41 | SteveTotaro | yes, i understand completely |
| 13:29.46 | [TK]D-Fender | SteveTotaro: I just typo'd it here, ig deal. |
| 13:29.54 | [TK]D-Fender | big* |
| 13:29.55 | Katty | [TK]D-Fender: http://www.youtube.com/watch?v=Axur5W83znw |
| 13:30.13 | SteveTotaro | i know you feel good correcting me, but i am correct |
| 13:30.18 | r0land | lol |
| 13:30.40 | SteveTotaro | but pretend you did so you can feel good about yourself, i will play along, ok? |
| 13:31.10 | r0land | one quest! currently with this: exten =>_01,1,Dial(SIP/$(EXTEN)@300) if i dial 01, i get a tone from 300! i tried changing "01" to "#1" but it didnt work! |
| 13:32.08 | r0land | ps: i guess [TK]D-Fender met his match :P |
| 13:32.27 | SteveTotaro | game, set, and match... |
| 13:32.46 | SteveTotaro | ;) |
| 13:33.01 | *** join/#asterisk w9sh (n=chatzill@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:33.07 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:33.07 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:33.12 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:33.20 | Katty | fskrotzki_: (= |
| 13:33.51 | *** part/#asterisk dwayne (i=dwayne@76.29.245.9) |
| 13:33.53 | fskrotzki | Morning Katty, how are we doing this fine day? |
| 13:34.23 | [TK]D-Fender | r0land: you want to use #1 instead of 01? add "pedantic=yes" under [general] in sip.conf |
| 13:34.32 | Katty | fskrotzki_: sleepy, but otherwise just ducky. how're you? |
| 13:34.57 | Katty | needs to clean her office. for Real. |
| 13:35.10 | fskrotzki | a bit more awake then you, but good. looking forward to a long weekend with no real plans.... |
| 13:35.25 | Katty | tell me about it!!! i'm already plotting. |
| 13:35.41 | Katty | maybe i'll wear PJs the entire weekend. |
| 13:35.56 | Katty | [TK]D-Fender: oh. i dreamt it was thanksgiving last night. |
| 13:36.03 | Katty | [TK]D-Fender: specifically about roasted turkey ^_- |
| 13:36.10 | Katty | [TK]D-Fender: except it came out more like bbqed turkey. |
| 13:36.16 | fskrotzki | hehe, I wish... (to both Katty and [TK]D-Fender)... |
| 13:37.15 | fskrotzki | boy scouts is picking up, just finished last week of camping and resident camp stuff, two weeks from today we start the popcorn drive for 2+ months... (thanksgiving is the end of my crazy season). |
| 13:37.21 | lmadsen | if I use Monitor(), and I have a call that goes: IAX2 --> Monitor() --> SIP phone, then I have files in.wav and out.wav.... does that mean the out.wav is the audio flowing from the SIP channel back to the IAX2 channel? |
| 13:37.53 | Qwell | lmadsen: IAX2 initiated the call? |
| 13:38.20 | Qwell | I would think IAX2>SIP=out, SIP>IAX2=in. don't quote me on that though |
| 13:38.33 | lmadsen | yes, the call comes from IAX2 into the box, and then the box sets up a Dial() to the SIP phone |
| 13:38.47 | lmadsen | hrmmm... I was thinking it was on the other side... |
| 13:38.51 | Qwell | oh, hmm |
| 13:38.59 | lmadsen | IAX2 --> Monitor() --> Dial(SIP/foo) --> SIP (foo) |
| 13:39.12 | lmadsen | in.wav <--- |
| 13:39.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:39.15 | lmadsen | errr |
| 13:39.19 | lmadsen | in.wav ---> |
| 13:39.23 | lmadsen | out.wav <--- |
| 13:39.46 | lmadsen | in is audio received from iax2, and out is audio recieved from sip |
| 13:40.24 | Qwell | damn perspective.. |
| 13:40.28 | lmadsen | heh :) |
| 13:40.29 | lmadsen | exactly |
| 13:40.50 | lmadsen | I'm thinking that because the Monitor() is called before DIal(), so it would only know of the IAX2 channel |
| 13:40.58 | Qwell | in and out seem like silly names, in that scenario |
| 13:41.02 | lmadsen | I agree |
| 13:41.08 | Qwell | both have audio flowing in and out |
| 13:41.12 | lmadsen | pre and post would be better..... |
| 13:41.17 | lmadsen | or something like that |
| 13:41.20 | lmadsen | left and right? :) |
| 13:41.33 | *** join/#asterisk thx_man (n=thx_man@port-83-236-207-89.static.qsc.de) |
| 13:45.07 | *** join/#asterisk underguiz (n=undergui@unaffiliated/underguiz) |
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| 14:22.02 | mgroman | Does hardware echo cancelation have any effect on phantom calls for tdm410 cards? |
| 14:23.37 | Corydon76-dig | mgroman: you might want to look at one of the other parameters for phantom calls, like ringdebounce |
| 14:24.09 | Corydon76-dig | Echocan only affects delivery of voice once the call is already up |
| 14:24.40 | mgroman | Corydon76-dig: The issue was resolved yesterday (zaptel svn), but russellb and Digium support were both asking me if the card had echo cancellation... just curious if that had any effect on the situation |
| 14:25.19 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:25.40 | Corydon76-dig | Hmmm |
| 14:26.09 | mgroman | And I guess for an IVR, EC is useless? |
| 14:26.29 | Corydon76-dig | EC is never useless for analog lines |
| 14:27.14 | Corydon76-dig | In fact, analog lines are the number one reason to have EC |
| 14:27.20 | [TK]D-Fender | s/analog lines/period/ |
| 14:27.25 | Corydon76-dig | Digital lines, less so |
| 14:27.53 | [TK]D-Fender | Corydon76-dig: After the nightmares I went through here at the start "less so" doesn't reassure me one bit :) |
| 14:28.06 | mgroman | because with digital lines, the transmission is either "there" or "not there", analog is more ambiguous? |
| 14:31.22 | M1s3ry | mgroman, more than likely Digium support would have asked so that we could make sure you were running the latest firmware for the card to fix any other issues that may come about later on. |
| 14:31.34 | Corydon76-dig | mgroman: with analog lines, you get echo from the 2-wire to 4-wire conversion |
| 14:32.12 | mgroman | ah |
| 14:32.59 | Corydon76-dig | On digital lines, the only way to get echo is the audio feedback between the speaker and microphone on the handset |
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| 14:46.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:48.09 | aliver | When someone dials an extension that doesn't match anything in the extensions.conf dialplan, does it go to the "i" extension? |
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| 14:51.34 | hsv-al | . |
| 14:52.12 | aliver | How can I find out what codec a remote SIP peer/friend supports? |
| 14:52.19 | aliver | codec(s) even. |
| 14:56.33 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 14:57.25 | [TK]D-Fender | aliver: in SIP certainly not. it just 404's |
| 14:57.48 | [TK]D-Fender | aliver: And go look at the SIP debug of a call attempt |
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| 15:00.38 | mgroman | i just dont like ekiga |
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| 15:05.42 | jeev | wow |
| 15:06.03 | jeev | i think spam got 500000 times worse, i gogt an email saying we have hijacked your baby, we want 50,000 us dollars for it. LOL with a zip file for photos |
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| 15:09.47 | Great_Anta_Baka | if i only have about 32 similtaneous calls max will the fact that i have raid-1 affect performance |
| 15:09.51 | Great_Anta_Baka | all calls will be sip |
| 15:10.31 | *** join/#asterisk ElSonico (n=tav@gprs-prointernet-ff476a00-16.dhcp.inet.fi) |
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| 15:14.28 | [TK]D-Fender | Great_Anta_Baka: And what does your HD have to do with call performance? |
| 15:15.10 | [TK]D-Fender | Great_Anta_Baka: Unless it hogs your CPU to death which would just be pathetic |
| 15:15.28 | aliver | [TK]D-Fender well, in the case I'm trying to figure out what codecs my outbound SIP provider (bandwidth.com) supports. I'll just try a call and see. |
| 15:16.37 | *** join/#asterisk sant0sk1 (n=sant0sk1@ip72-206-113-224.om.om.cox.net) |
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| 15:18.46 | aliver | Does G726 use compression or is it all framing and jitter correction advantages over G711 ? |
| 15:19.13 | aliver | The thing is, I'm having problems with outbound audio quality. |
| 15:19.40 | aliver | I don't have G729 available, so I'm trying to shoot for the next best thing. |
| 15:19.42 | aliver | any suggestions? |
| 15:19.53 | jblack | Yeah. ulaw. |
| 15:21.13 | aliver | Isn't ulaw just raw, uncompressed, and bandwidth-hogging? I'd think iLBC would at least be better. |
| 15:21.30 | aliver | Just asking, I don't really know, though. |
| 15:21.58 | [TK]D-Fender | aliver: What do they suppor? |
| 15:22.00 | jblack | Ok, well, gsm will cost you 8KB/sec per call and is comperable to pstn. That's pretty cheap. |
| 15:22.06 | [TK]D-Fender | support* |
| 15:22.47 | [TK]D-Fender | aliver: Not compressed, just half the size due to direct sampling rate. |
| 15:22.53 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 15:22.59 | aliver | [TK]D-Fender It appears they support all the G7XX codecs, but not GSM or iLBC as far as I can tell. It's hard to tell from the sip debug output what is the phone and what is the phone provider in when it talks about codecs. |
| 15:23.19 | aliver | Hmm, well, I'll have to call as ask if they'll allow GSM. |
| 15:23.23 | [TK]D-Fender | aliver: Yes its perfectly easy to see. |
| 15:23.28 | [TK]D-Fender | aliver: pastebin that call attempt |
| 15:23.39 | [TK]D-Fender | aliver: And you'd have SEEN if they did |
| 15:24.07 | [TK]D-Fender | aliver: Since you're having trouble with the big print, pastebin it so I can simply show you. |
| 15:26.38 | jblack | oh cool. Gustav is heading towards new orleans |
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| 15:30.04 | sant0sk1 | i have asterisk setup on public ip receiving sip trunked DID calls and passing them to adhearsion. When making a call I get the following asterisk error: "chan_sip.c:14035 handle_request_invite: Call from '' to extension '4022184044' rejected because extension not found." you can see my asterisk configs at: http://gist.github.com/7281 any help is much appreciated! |
| 15:30.17 | delphus | is there anyway to set g711 gain over h323 ? |
| 15:30.45 | Qwell | delphus: I think in 1.6 you can |
| 15:30.49 | Qwell | func_volume |
| 15:31.08 | delphus | Qwell: 1.6 wont help me ... |
| 15:31.09 | Qwell | maybe that only works on zap though. who knows |
| 15:31.59 | delphus | Qwell: well thanks anyway ;) |
| 15:33.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:34.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:35.44 | [TK]D-Fender | sant0sk1: Go look at the SIP debug for your failed call. |
| 15:36.31 | [TK]D-Fender | Qwell: From what I heard its channel-agnostic |
| 15:37.01 | sant0sk1 | [TK]D-Fender: /var/log/asterisk ? |
| 15:37.14 | [TK]D-Fender | sant0sk1: no, * CLI with SIP debug enabled |
| 15:37.23 | Qwell | hmm, I'm thinking of something else then. |
| 15:38.34 | [TK]D-Fender | Qwell: I know rxgain/txgain work pretty well for zap ;) |
| 15:38.37 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:39.58 | aliver | http://pastebin.com/d106088e1 <--- this seems to indicate all they support is ulaw |
| 15:40.03 | aliver | but they claim to support g729 |
| 15:40.09 | aliver | and all G7XX |
| 15:40.21 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:40.46 | [TK]D-Fender | aliver: can you try that frmo the START of the call... |
| 15:42.06 | *** join/#asterisk n3hxs (n=HAMming@c-71-57-230-234.hsd1.pa.comcast.net) |
| 15:42.26 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-217a0953472c08bd) |
| 15:43.14 | sant0sk1 | [TK]D-Fender: found the problem. thank you |
| 15:49.23 | mattx86 | any ideas why my zap groups aren't working? console output and zapata.conf are here: http://pastebin.com/d1ff64458 |
| 15:49.34 | *** part/#asterisk sant0sk1 (n=sant0sk1@ip72-206-113-224.om.om.cox.net) |
| 15:51.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:52.33 | [TK]D-Fender | mattx86: you must define multiple groups on *1* line. You do not to this over mutliple. |
| 15:52.37 | [TK]D-Fender | multiple. |
| 15:52.53 | mattx86 | ah. I'll change it right quick |
| 15:53.04 | [TK]D-Fender | mattx86: group=2,3 |
| 15:54.38 | aliver | http://pastebin.com/m40520591 <-- all the sip debug info. |
| 15:54.48 | mattx86 | [TK]D-Fender: nice, thanks! |
| 15:54.54 | aliver | Can you figure out what codecs they support? |
| 15:55.00 | Great_Anta_Baka | if i make a call from an analogue line i cant interact with the ivr but whenever i call from an isdn line or a mobile phone it works |
| 15:55.10 | aliver | <PROTECTED> |
| 15:55.17 | Great_Anta_Baka | what could the possible solutions to this problem be? |
| 15:55.18 | aliver | So, I don't see how to identify them. |
| 15:55.31 | *** join/#asterisk Levonk (n=lk@adsl-76-237-15-132.dsl.lsan03.sbcglobal.net) |
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| 15:57.06 | [TK]D-Fender | aliver: I'm not wading thought ^%$# 3000 lines of crap. debug the PEER and look at an INCOMING call. |
| 15:59.31 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 16:00.30 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:01.06 | Katty | wibbles |
| 16:02.51 | Great_Anta_Baka | if i make a call from an analog line i cant interact with a specific IVR but whenever i call from an isdn line or a mobile phone it works |
| 16:03.00 | mgroman | What might cause an app from registering with Asterisk? not being executable, not being in default app folder, not being loaded by modules.conf... is there anything else to check? |
| 16:04.10 | Katty | wobbles |
| 16:05.55 | aliver | How can I turn off 'sip debug' without restarting asterisk? |
| 16:06.24 | aliver | and don't tell me 'set debug 0' because that doesn't AT ALL. |
| 16:07.20 | rivalmel | sip set debug off |
| 16:08.14 | *** join/#asterisk zdraper (n=zdraper@64-142-43-180.dsl.static.sonic.net) |
| 16:10.01 | *** join/#asterisk bjwebb (n=bjwebb@tuxhacker/Bjwebb) |
| 16:10.35 | aliver | mercury*CLI> sip set debug off |
| 16:10.35 | aliver | No such command 'sip set' (type 'help' for help) |
| 16:10.38 | aliver | wrong |
| 16:11.03 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 16:11.25 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 16:12.17 | jblack | Actually, sip set debug off is how one turns off sip debug. |
| 16:12.22 | jblack | There's something wrong with your system. |
| 16:12.34 | aliver | Like for example it's not running 1.6 beta or something? |
| 16:12.37 | aliver | It's 1.2 |
| 16:13.53 | aliver | It's "sip no debug" |
| 16:14.08 | aliver | finally. |
| 16:16.16 | Paige_ | any idea why asterisk 1.6 would be looking for gtk+ when compiling with --imap? |
| 16:16.44 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 16:16.56 | Qwell | Paige_: it'll always look for it - but it doesn't require it |
| 16:16.59 | Qwell | is it failing or something? |
| 16:17.10 | Paige_ | configure is failing |
| 16:17.17 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:17.21 | Qwell | can you pastebin the output? |
| 16:17.22 | Qwell | ~pb |
| 16:17.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:17.47 | Qwell | actually, put your config.log up there |
| 16:18.38 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:20.54 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 16:21.00 | mgroman | i always pipe to `less` rather than `more`, am i a pessimist? |
| 16:21.15 | Qwell | no, more sucks |
| 16:22.00 | jeev | damn dood |
| 16:22.03 | jeev | dr dre's son is dead |
| 16:22.32 | mgroman | xzhibits son died a month ago or so |
| 16:22.40 | Sargun | Who? |
| 16:22.50 | jeev | really mgroman? i didn't know.. wow |
| 16:22.58 | Sargun | Who is mgroman? |
| 16:23.02 | Sargun | err |
| 16:23.06 | mgroman | lol |
| 16:23.07 | Sargun | xzhibit? |
| 16:23.09 | jblack | marijuana grow man ? |
| 16:23.20 | mgroman | rapper/host of hit show on mtv "pimp my ride" |
| 16:23.31 | jeev | he was a really good guy too |
| 16:23.33 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 16:23.43 | Sargun | oh |
| 16:23.47 | jeev | i dont see that anywhere about his son |
| 16:24.06 | mgroman | jeev: maybe it was further back then...i read about it a month ago |
| 16:24.12 | jeev | dre is my my good friends friend.. they live a few miles apart, i told him and my friend is sad now.. they hadn't spoke for a while |
| 16:24.17 | jeev | ah dunno |
| 16:24.58 | jeev | ahh, infant son mgroman.. |
| 16:25.01 | Paige_ | Qwell, http://pastebin.ca/1185399 |
| 16:25.50 | Qwell | cygnus? |
| 16:26.18 | Paige_ | what about it |
| 16:26.25 | Paige_ | used to work for them years ago |
| 16:26.36 | Qwell | ahh, okay,I thought you were doing something silly like installing on cygwin.. |
| 16:26.48 | Paige_ | nope |
| 16:26.48 | Qwell | no, it's failing checking for IMAP |
| 16:26.56 | [TK]D-Fender | qwe No, that was baliktad ;) |
| 16:27.30 | *** part/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 16:28.01 | Paige_ | the error was something about IMAP_TK |
| 16:29.03 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:31.03 | mikealeonetti | okay, so I'm part of a business that is interested in going to VoIP. I'm just interested in what we need as far as obtaining lots of phone lines. Should I just go looking for a T1 company? |
| 16:31.40 | [TK]D-Fender | mikealeonetti: ... can you please clarify that. a LOT. |
| 16:31.49 | mikealeonetti | lol |
| 16:32.07 | jeev | lol |
| 16:32.34 | jeev | mikealeonetti, fender just acts that way but underneath it all, he loves you more than your own mother and father |
| 16:32.57 | [TK]D-Fender | jeev: c'mon, they can't possibly hate him THAT much! |
| 16:33.04 | mikealeonetti | oh they do |
| 16:33.07 | [TK]D-Fender | :D |
| 16:33.31 | mikealeonetti | it's amazing what toll having a son in prison takes on young parents |
| 16:33.45 | mikealeonetti | I wish they didn't stop paying for therapy |
| 16:33.59 | jeev | wow |
| 16:34.07 | jblack | Whatever toll it is, it's not enough. |
| 16:34.13 | mikealeonetti | lol |
| 16:34.17 | [TK]D-Fender | the rapist <- fucks with your mind |
| 16:34.26 | mikealeonetti | ever see "In Treatment?" |
| 16:34.28 | mikealeonetti | <3 that sow |
| 16:34.29 | mikealeonetti | show* |
| 16:35.05 | mikealeonetti | every 5 minute Gabriel screams "THERE IS NO KEIZER SOZE!" |
| 16:36.35 | mikealeonetti | I am interested in putting together an Asterisk server, buying a bunch of hardware, and setting up an intercompany VoIP service. We need some phone lines. Should I look for a T1 company for the phone lines, or go with a local VoIP->analog phone company deal? |
| 16:37.14 | [TK]D-Fender | mikealeonetti: What does "VoIP" have to do with "lines"? |
| 16:37.24 | mikealeonetti | we need to dial out |
| 16:37.31 | mikealeonetti | is what I mean |
| 16:37.40 | mikealeonetti | so we need some phone numbers, I guess |
| 16:37.53 | [TK]D-Fender | mikealeonetti: Ok, and if you go for normal "lines" (via a hardware card, etc), then where does "VoIP" come in? |
| 16:38.29 | mgroman | Is darren sessions in here? |
| 16:38.43 | Paige_ | so any suggestions on fix ing my error? |
| 16:38.46 | mikealeonetti | [TK]D-Fender: normal "lines" as in analog? |
| 16:39.25 | [TK]D-Fender | mikealeonetti: Analog, T1, etc. |
| 16:40.09 | mikealeonetti | [TK]D-Fender: exactly |
| 16:40.37 | [TK]D-Fender | mikealeonetti: Exactly what? What is "VoIP" with you are dealing with PHYSICAL lines? |
| 16:40.48 | *** join/#asterisk Levonk (n=lk@adsl-75-62-139-107.dsl.lsan03.sbcglobal.net) |
| 16:41.23 | mikealeonetti | [TK]D-Fender: well, is it worth purchasing a T1? |
| 16:42.20 | [TK]D-Fender | mikealeonetti: You don't seem to know what you want. |
| 16:42.48 | mikealeonetti | [TK]D-Fender: exactly. unfortunately the rapist doesn't help with that one. |
| 16:43.16 | [TK]D-Fender | mikealeonetti: Sorry, we don't sell clues here. |
| 16:43.44 | mikealeonetti | [TK]D-Fender: you do help people, though, right? |
| 16:44.07 | mikealeonetti | I mean, jeev just told me you loved me |
| 16:44.10 | Paige_ | omfg some people are such idiots |
| 16:44.17 | [TK]D-Fender | mikealeonetti: God helps those who help themselves. |
| 16:44.27 | [TK]D-Fender | mikealeonetti: No, he jsut said I loved you more than your PARENTS <- |
| 16:44.52 | Paige_ | [intra]lanman, you seeing this shit? |
| 16:44.54 | [TK]D-Fender | Paige_: You gain wisdom child. |
| 16:45.10 | [intra]lanman | what shit? |
| 16:46.01 | mgroman | yawn |
| 16:46.01 | Paige_ | n/m go back to sleep :) |
| 16:46.01 | Paige_ | qwell, any idea how to resolve my configuration issue? |
| 16:46.09 | Qwell | Paige_: properly install the imap toolkit stuff. |
| 16:46.10 | mgroman | why do shared objects get named like so --> "libthing.so.3", why is that number joined on the end? |
| 16:46.16 | Paige_ | it is |
| 16:46.30 | Qwell | mgroman: it's the soversion |
| 16:46.30 | [TK]D-Fender | mgroman: Version. |
| 16:46.36 | [intra]lanman | dozes off as per Paige_'s suggestion |
| 16:46.45 | mgroman | ahh thanks |
| 16:48.25 | mikealeonetti | alright, you win. cheers then. |
| 16:48.27 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:48.28 | aliver | When I transfer a user to voicemail they get the "temp" message always. Is that normal? |
| 16:48.46 | aliver | Shouldn't they get the unavailable message? |
| 16:49.16 | [TK]D-Fender | aliver: For as long as you leave it there, YES |
| 16:49.30 | *** join/#asterisk jameswf-home (n=james@cl-24.phx-01.us.sixxs.net) |
| 16:50.42 | aliver | Well, when they don't have a temp message it just plays 'vm-intro' (the generic message) instead of their unavailable message. |
| 16:50.55 | aliver | Do I need to add the "u" on the end of Voicemail() ? |
| 16:51.18 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 16:52.12 | aliver | I guess I need Voicemail(u${EXTEN}) |
| 16:58.46 | aliver | When you do a 'sip show channels' is that showing calls that are actually in progress or does it include other states as well? |
| 17:00.31 | jameswf-home | show channels shows active calls |
| 17:00.40 | jameswf-home | sip show channels shows sip channels |
| 17:00.51 | jameswf-home | *add core as needed |
| 17:02.41 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:04.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:05.46 | aliver | Can someone help me figure out what this line is doing ? |
| 17:05.48 | aliver | exten => s,3,GotoIfTime(08:00-17:00,mon-fri,*,*?6) |
| 17:06.01 | aliver | What is the last argument "*?6" doing? |
| 17:06.06 | bjwebb | apache keeps dying on me :( |
| 17:07.00 | jeev | Fender, for the big office, since i have all the phones speak with 192.168.0.1, what's the best way for it to determine if the primary * box is down, to go out the second one ? |
| 17:11.43 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:12.33 | aliver | What is the last argument "*?6" doing? |
| 17:13.12 | mvanbaak | if it matches it will jump to priority 6 |
| 17:13.24 | aliver | thanks. |
| 17:13.44 | mvanbaak | have a look at: 'core show application GotoIfTime' |
| 17:14.19 | aliver | What does this do ? "${EXTEN}"]?3:5 |
| 17:14.31 | aliver | at least the ?3:5 part |
| 17:14.41 | aliver | if it matches go to priority five? |
| 17:15.07 | aliver | The whole line reads: exten => 700,2,GotoIf($["${CALLERID(num)}" = "${EXTEN}"]?3:5) |
| 17:15.08 | mvanbaak | no, if it matches go to 3, otherwise go to 5 |
| 17:15.20 | aliver | Ahh, okay. Thanks. that helps a lot to know that syntax. |
| 17:15.36 | *** join/#asterisk mpruett (n=mpruett@24-240-203-84.static.stls.mo.charter.com) |
| 17:15.41 | aliver | I wish I could find a cheat-sheet for that syntax online. |
| 17:15.47 | mvanbaak | <test>?when_true:when_false |
| 17:15.56 | aliver | Ah ha. |
| 17:16.35 | mvanbaak | food |
| 17:16.40 | *** join/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
| 17:17.25 | nny_2 | any DCAP certified peeps know of study material outside of the book? Gonna take the test here soon, but the Boot Camp course is a bit outside of my budget :) |
| 17:18.13 | *** join/#asterisk JenniferAkemi (n=akemi@MTLXPQAK-1178074603.sdsl.bell.ca) |
| 17:18.39 | mpruett | Guys what is the URL of the post bin used to post code into> |
| 17:18.49 | *** join/#asterisk jameswf-home (n=james@cl-24.phx-01.us.sixxs.net) |
| 17:19.09 | nny_2 | ~pb |
| 17:19.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:19.24 | mpruett | Thanks! |
| 17:19.31 | [TK]D-Fender | aliver: "core show application gotoif" , "core show application gotoiftime <- next time just read the INSTRUCTIONS." |
| 17:20.15 | [TK]D-Fender | aliver: For as long as you leave it there, YES <------- (temp overrides ALL OTHERS) |
| 17:20.33 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-88773e9828271b25) |
| 17:20.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:25.32 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:26.11 | jeev | shouldn't ./configure --prefix=/usr/local/asterisk place everything in there when make installing? it's not! |
| 17:26.37 | mgroman | jeev: where is it putting it and what is it putting there |
| 17:27.51 | jeev | bin include lib sbin share, that's all it puts. it still puts etc in /etc |
| 17:27.53 | jeev | damn linux. |
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| 17:30.40 | jeev | jeffgus, i used to own zimage.org |
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| 17:54.22 | jeffgus | jeev, oh? you're the guy eh? |
| 17:54.43 | jeffgus | i've had zimage.com since 1997 |
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| 17:59.20 | *** join/#asterisk techman97 (n=myweiner@97-91-103-181.dhcp.roch.mn.charter.com) |
| 18:00.29 | techman97 | hello all, I'm seeing some weird activity on my asterisk box....looking at the full log, it looks like I have registrations going on to my SIP providers and phones every 30ms? Any hints? |
| 18:01.42 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:01.49 | lmadsen | change the registration timeout in sip.conf to not be 30ms |
| 18:02.09 | lmadsen | unless the other side is of course requesting 30ms, in which case asterisk will probably follow what the other side is telling your box |
| 18:02.19 | [TK]D-Fender | 30ms is INSANE |
| 18:02.24 | lmadsen | aye |
| 18:02.31 | [TK]D-Fender | Heck expecing a 30ms PING is pretty much insane |
| 18:02.42 | techman97 | yeah, tell me about it |
| 18:02.53 | techman97 | PBX has been working great for 2 years, minus some ISP issues |
| 18:03.16 | techman97 | and for the last week (no updates or anything applied for months) we've been having to reboot the box here and there, reboot the SIP phones... |
| 18:03.23 | techman97 | so I looked in the logs... |
| 18:03.41 | lmadsen | minexpiry=60 |
| 18:03.41 | lmadsen | defaultexpiry=1200 |
| 18:04.57 | techman97 | all debug messages I'm seeing - no warnings except for the usual crap. I am seeing a FLOOD of "testing address...", "Scheduled a registration timeout", "Stopping retransmissions on <ip address> Match Found", "Registration Successful", "CAncelling Timeout". Then, a few milliseconds later, again the same stuff repeats |
| 18:05.01 | techman97 | over and over and over |
| 18:05.07 | techman97 | (I'm paraphrasing the logs) |
| 18:05.29 | techman97 | box is on a private network, SIP / NAT to external phones. Been working great for 2 years as I said |
| 18:05.33 | techman97 | no idea. |
| 18:06.00 | techman97 | we moved the * box to another ISP thinking it was a bad DSL line or something forcing the phones to lose connection and reregister... |
| 18:16.15 | scooby2 | this is probably a dumb question, but how to you show outbound calls? zap show channels shows inbound |
| 18:16.57 | [TK]D-Fender | scooby2: "show channels |
| 18:17.07 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 18:17.26 | scooby2 | thank you |
| 18:18.17 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 18:21.52 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 18:27.39 | techman97 | aahh ha! |
| 18:27.54 | techman97 | I have a bad NIC or switch...the error count on my eth0 is through the roof |
| 18:27.56 | techman97 | damn hardware LOL |
| 18:30.24 | *** join/#asterisk vonkleist (n=gcontrer@189.155.100.170) |
| 18:30.32 | vonkleist | Hi everybody |
| 18:30.40 | techman97 | hi vonkleist |
| 18:30.52 | vonkleist | Is there any way to set a "dialing password" ? |
| 18:30.55 | vonkleist | I mean... |
| 18:31.01 | vonkleist | there are some restriction in a dialplan |
| 18:31.09 | [TK]D-Fender | vonkleist: Thats it |
| 18:31.12 | vonkleist | but want some users to override that restrictions |
| 18:31.19 | vonkleist | dialing in some kind of password |
| 18:31.25 | [TK]D-Fender | vonkleist: Its your dialplan, do whatever you want |
| 18:32.02 | vonkleist | well... I can set an "extension" with a password included on it, but then when this user dial their password on the phone, the phone will record the password too... |
| 18:32.42 | vonkleist | and if the restricted user use the redial button on its hard phone, he will see the password... |
| 18:32.51 | ghenry | with zaptel.conf can you leave spans unconfigured? |
| 18:33.03 | [TK]D-Fender | vonkleist: then don't make it PART of the dialed extension. PROMPT him for a password |
| 18:33.09 | [TK]D-Fender | ghenry: yes |
| 18:33.20 | vonkleist | how do I do that? |
| 18:33.20 | ghenry | we have 4 spans and only want 3&4 |
| 18:33.50 | techman97 | ghenry: yes |
| 18:34.17 | [TK]D-Fender | vonkleist: "core show application read" , "core show application authenticate", etc.... take you pick. And while your'e at it, go read the entire application list. |
| 18:34.21 | techman97 | ghenry: I do dual span cards and actually put the unused span in it's own group |
| 18:34.26 | ghenry | when we comment out span 1& 2 zaptel crashes on startup |
| 18:34.28 | vonkleist | ok, let me check |
| 18:34.43 | ghenry | can yo configure a dummy span? |
| 18:34.48 | ghenry | so it doesn't do anything |
| 18:35.08 | ghenry | http://nopaste.com/p/ab7x4fXnZ |
| 18:35.19 | ghenry | when we comment out span 1&2 and settings for it |
| 18:35.22 | vonkleist | [TK]D-Fender, wow... |
| 18:35.25 | ghenry | reboot fails |
| 18:35.31 | ghenry | loading zaptel |
| 18:35.40 | ghenry | 3&4 are right settings, as they load fine |
| 18:36.14 | *** join/#asterisk jameswf-home (n=james@cl-24.phx-01.us.sixxs.net) |
| 18:36.23 | [TK]D-Fender | ghenry: If you only want 3 & 4 then why aren't 1 & 2 commented out? |
| 18:36.39 | ghenry | as /etc/init.d/zaptel restart doesn't work |
| 18:37.35 | [TK]D-Fender | ghenry: comment the other spans & their associated channel out, and retest "ztcfg -vvvv" manually yourself |
| 18:37.53 | ghenry | other spans? |
| 18:37.56 | ghenry | 1&2? |
| 18:38.01 | ghenry | or all of them? |
| 18:38.24 | *** join/#asterisk mattx86 (n=matt@static2073.uctnwd.ken-tennwireless.com) |
| 18:38.45 | [TK]D-Fender | \gheThe ones you clearly don't want to use |
| 18:39.07 | ghenry | ok |
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| 18:40.00 | mattx86 | is there anything special to do for a POTS line + Channel Bank setup, where the inbound/outbound telco lines won't hang up? so far I've not had success with fxs ks/ls signalling. I'm now trying fxsgs |
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| 18:44.15 | [TK]D-Fender | mattx86: Make sure your telco has enabled CDS |
| 18:44.19 | [TK]D-Fender | ~cds |
| 18:44.20 | jbot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 18:44.42 | kaldemar | mattx86: busydetect=yes may also help with fxsks, in case you don't have it. |
| 18:45.09 | mattx86 | [TK]D-Fender: is this something that's non-standard (on a US POTS line)? |
| 18:45.27 | [TK]D-Fender | mattx86: its a service you ahve to ask for. |
| 18:45.37 | [TK]D-Fender | kaldemar: EW. |
| 18:45.49 | Kobaz | heh |
| 18:46.01 | mattx86 | kaldemar: yeah, tried that too |
| 18:46.09 | kaldemar | [TK]D-Fender: it's not that disgusting, especially if it gets the job done. :) |
| 18:49.45 | *** part/#asterisk red1 (n=red1@158.65.49.60.klj04-home.tm.net.my) |
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| 18:59.52 | kfife | Hi guys. I've been steeped in Asterisk for about two years, but I"m compilign Zaptel for the first time: supposedly relies on a package called kernel-smp-devel, but my repository says there's no such package. Does it go by another name? |
| 19:00.33 | C4away | kernel-devel should install the smp version if that is what you are running |
| 19:01.10 | kfife | Thank you. That explains it. I'm running a dual pentium system. I'm assuming SMP stands for Symmetric Multiprocessing? |
| 19:02.33 | *** join/#asterisk JenniferAkemi (n=akemi@MTLXPQAK-1178074603.sdsl.bell.ca) |
| 19:03.21 | tzafrir_laptop | kfife, what kernel version do you use? |
| 19:03.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.55 | tzafrir_laptop | As of kernel 2.6.17, all kernels are SMP (mostly. optimizations are patched at runtime) |
| 19:04.46 | kfife | tzafrir_laptop: Linux asterisk 2.6.18-92.1.10.el5 #1 SMP Wed Jul 23 03:55:54 EDT 2008 i686 i686 i386 GNU/Linux |
| 19:05.16 | kfife | tzafrir_laptop: so that's SMP ala dual core etc. |
| 19:05.41 | *** join/#asterisk [gnubie] (n=gnubie@203.177.180.52) |
| 19:05.51 | kfife | tzafrir_laptop: as in: it's becoming impossible to find a single core processor these days? |
| 19:06.41 | mattx86 | [TK]D-Fender: hm.. the AT&T Repair lady said they couldn't provide that service |
| 19:07.03 | [TK]D-Fender | mattx86: Get a better opinion |
| 19:08.02 | mattx86 | heh. first, she said the number I gave her was through a wireless carrier and not a land line. |
| 19:08.28 | mattx86 | I'll try it again in a little bit. |
| 19:08.57 | tzafrir_laptop | kfife, if there's just one CPU, the kernel will patch itself at boot accordingly |
| 19:08.59 | kfife | mattx86: When dealing with the phone company, I've had to call back 4 or 5 times to get services that I KNOW they have, but the first 4 peoplel say "We can't do that" or "That's impossible" when really they mean "I DON'T KNOW". In the end, you can get it if it's offerred |
| 19:09.18 | [TK]D-Fender | mattx86: So they're incompetant ANY lying. |
| 19:09.22 | [TK]D-Fender | AND* |
| 19:10.05 | mattx86 | I gotcha. :) |
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| 19:40.50 | ghenry | TE410P and Centos don't like |
| 19:41.04 | ghenry | each other |
| 19:41.09 | [netman] | why not? |
| 19:41.24 | ghenry | /etc/init.d/zaptel start hangs the box |
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| 19:41.46 | ghenry | why would zaptel start if chkconfig zaptel off is run? |
| 19:41.56 | ghenry | i.e. modules being loaded |
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| 19:43.35 | ghenry | why would the zaptel module be loaded if /etc/init.d/zaptel is off? |
| 19:44.28 | *** join/#asterisk haryv (n=hary@S0106001d7e52cc78.vs.shawcable.net) |
| 19:44.48 | Corydon76-dig | ghenry: if you probed a module that had zaptel as a dependency |
| 19:44.56 | ghenry | ah, ok |
| 19:45.01 | ghenry | how can I cehck that? |
| 19:45.02 | Corydon76-dig | ghenry: check /etc/modules and your /etc/modprobe.d directory |
| 19:45.42 | [gnubie] | weird |
| 19:45.57 | ghenry | crc_ccitt |
| 19:45.59 | ghenry | uses it |
| 19:46.04 | ghenry | how to stop that on startup for testing |
| 19:46.05 | [gnubie] | # /etc/init.d/zaptel start |
| 19:46.06 | [gnubie] | Zaptel telephony kernel driver: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 19:46.39 | Corydon76-dig | [gnubie]: check your /etc/zaptel.conf |
| 19:46.58 | Corydon76-dig | [gnubie]: are the modules loading correctly? |
| 19:47.44 | [gnubie] | Corydon76-dig: yes.. and i have fxo modules here and on my /etc/zaptel.conf i got a line that says: fxsks=1 |
| 19:48.05 | [TK]D-Fender | [gnubie]: modprobe wctdm |
| 19:48.14 | aliver | exten => s,25,Dial(SIP/731&SIP/733&SIP/734&SIP/735,40,wA(custom/11_vnet_ts_call_announce)) |
| 19:48.16 | [TK]D-Fender | [gnubie]: then redo "ztcfg -vvvv" |
| 19:48.20 | aliver | what is that wA stuff at the end? |
| 19:48.21 | *** join/#asterisk [tasty]freeze (n=derek@pool-71-171-23-53.nwrknj.east.verizon.net) |
| 19:48.27 | [gnubie] | Corydon76-dig and [TK]D-Fender: it's here => http://paste.debian.net/15648/ |
| 19:48.33 | [TK]D-Fender | aliver: "core show application dial" <- |
| 19:48.39 | aliver | okay |
| 19:49.46 | aliver | that's handy |
| 19:49.51 | [tasty]freeze | I know daisy chaining a incoming phone line to another build is bad, but is there a repeater or some solution I can use without digging up the ground to restore the normal voltage/current to the line as I am receiving a minor echo in the line using VoIP equipment. |
| 19:49.58 | *** join/#asterisk thx_man (n=thx_man@port-83-236-207-89.static.qsc.de) |
| 19:50.00 | ghenry | so why does crc_ccitt load up zaptel? |
| 19:50.56 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 19:51.39 | Corydon76-dig | [gnubie]: if you're loading wctdm, you shouldn't be loading ztdummy |
| 19:51.51 | [gnubie] | Corydon76-dig and [TK]D-Fender: kindly check this => http://paste.debian.net/15650/ |
| 19:52.36 | [TK]D-Fender | [gnubie]: And the modprobe I asked for? And unload ztdummy as Corydon76-dig suggested |
| 19:53.32 | *** part/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
| 19:53.45 | Corydon76-dig | You may have wctdm loaded, but /proc/interrupts shows that it's not taking interrupts |
| 19:54.08 | Corydon76-dig | [gnubie]: so you need to talk to the reseller from whom you purchased that card, for a possible RMA |
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| 19:55.59 | Corydon76-dig | [gnubie]: btw, the current cards from Digium no longer use the TigerJet interface |
| 19:56.26 | Corydon76-dig | and getting a newer card may help the situation |
| 19:57.27 | [gnubie] | ok |
| 19:57.56 | [gnubie] | i removed wctdm here.. totally, even the zaptel module is not there already |
| 19:59.59 | [gnubie] | i also tried unloading ztdummy and modprobing wctdm but still with no luck |
| 20:00.44 | Corydon76-dig | I'd say you have a dead card |
| 20:01.00 | [gnubie] | what is the wcopenpci ? |
| 20:01.09 | Corydon76-dig | No idea |
| 20:01.27 | Corydon76-dig | Is this a clone card? |
| 20:01.34 | [gnubie] | from openvox |
| 20:01.43 | Corydon76-dig | Yeah, there's the problem |
| 20:01.59 | Corydon76-dig | Good luck getting support from them... or even a return |
| 20:02.00 | [gnubie] | well, this is not brand new from them.. a friend gave me this card |
| 20:02.28 | *** part/#asterisk andylockran (n=andylock@genesis.zrmt.com) |
| 20:02.42 | [gnubie] | it used to work but it took months for me before i installed it on my pc |
| 20:03.05 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:03.22 | [gnubie] | and now, i was surprised that the leds are not blinking |
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| 20:05.42 | [gnubie] | another question.. trying to call an echo test and plays a file, i can only hear a choppy sound |
| 20:06.05 | [gnubie] | why is that so? the audio files that i have are all in gsm format |
| 20:07.16 | lowtek | [gnubie] What version of GCC did you compile * with? |
| 20:07.55 | [gnubie] | lowtek: gcc-4.2 on ubuntu 8.0.4.1 |
| 20:08.05 | lowtek | ~gcc |
| 20:08.06 | jbot | it has been said that gcc is the GNU Compiler Collection, http://gcc.gnu.org/ |
| 20:08.41 | lowtek | [gnubie]: recompile with GCC < 4.2 (i.e., 4.1) and that should solve it. |
| 20:09.31 | [gnubie] | lowtek: you mean, there is a problem in compiling asterisk and zaptel using gcc-4.2? |
| 20:09.40 | zoid_99 | that's what he means |
| 20:10.17 | zoid_99 | we rN INTO THE SAME THING |
| 20:10.18 | lowtek | [gnubie]: According to google and various in-channel chatter, yes. I've never tried it personally. |
| 20:10.20 | zoid_99 | oops |
| 20:10.32 | *** join/#asterisk Levonk (n=lk@adsl-76-227-119-76.dsl.lsan03.sbcglobal.net) |
| 20:10.41 | *** join/#asterisk metfan2007 (n=jc@189.135.238.90) |
| 20:10.46 | metfan2007 | hi all!!! |
| 20:10.52 | [gnubie] | i see.. |
| 20:11.13 | metfan2007 | Is there anyway to know if a pin of a Digium card is in T1 or E1 mode? via linux console? |
| 20:12.05 | [TK]D-Fender | metfan2007: "This card is best viewed with.... YOUR EYES" |
| 20:12.54 | metfan2007 | [TK]D-Fender: Hehehehe, I know it! but at the time I have not access to that box :S |
| 20:13.26 | [TK]D-Fender | metfan2007: Go ask someone who does. It should never have been installed without being checked out first. |
| 20:14.25 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) |
| 20:15.00 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 20:15.01 | metfan2007 | [TK]D-Fender: There's nobody at place... is in other city, in a empty house :S |
| 20:15.22 | [TK]D-Fender | metfan2007: Good luck... |
| 20:16.12 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
| 20:16.38 | metfan2007 | [TK]D-Fender: :S I know that there's a way to override the pin selection via software, is that correct? |
| 20:16.43 | ghenry | why is crc_ccitt loading zaptel module without /etc/init.d/zaptel loading up |
| 20:16.56 | LemensTS | What do you recommend for a doctor/dentist office program to notify patients of appointments? i figured there was something wrote already for this |
| 20:17.37 | [TK]D-Fender | metfan2007: if there was, why the hell would there be a physical JUMPER on the card? |
| 20:17.56 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 20:18.06 | metfan2007 | [TK]D-Fender: Hey.... take it easy... is only a question! :P |
| 20:18.24 | [TK]D-Fender | metfan2007: Yes, a remarkably silly one. |
| 20:21.36 | mchou | LemensTS: the issue is text to speech synthesis. Festival is not really acceptable unless you no longe want to practice. :) |
| 20:21.37 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:21.49 | mchou | longer* |
| 20:22.00 | metfan2007 | [TK]D-Fender: insmod wct4xxp t1e1override=0x00 |
| 20:22.06 | metfan2007 | [TK]D-Fender: Thanks anyway |
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| 20:23.06 | mchou | LemensTS: every other piece of the technology exists in asterisk. Just the text to speech synthesis falls down badly |
| 20:23.31 | kfife | I heard something a while ago about problems compilign zaptel 1.4.11 on CENTOS 5. Now I'm compiling zap on CENTOS 5 and running into a compile issue. Anybody heard of this Here's the error: http://pastebin.com/m7c490fee |
| 20:23.52 | jblack | ~centos5bug |
| 20:23.55 | Qwell | kfife: see topic |
| 20:24.07 | jblack | ~centos52bug |
| 20:24.07 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 20:24.12 | [TK]D-Fender | Seriously... how bibg do I have to make that notice? |
| 20:24.19 | [TK]D-Fender | big* |
| 20:24.35 | jblack | Really want an answer to that? |
| 20:24.46 | [TK]D-Fender | jblack: I'm thinking flashing neon. |
| 20:24.56 | mchou | LemensTS: if you can find another text to speech synthesis engine (on linux) that generates acceptable speech then you are home free |
| 20:25.08 | jblack | Not good enough. A lot of americans still think their army is in iraq to catch al qaeda. |
| 20:25.10 | [TK]D-Fender | Tovarishch: Dosvedania |
| 20:25.11 | kfife | Qwell: Thanks |
| 20:25.31 | Tovarishch | [TK]D-Fender: Dosvedania |
| 20:26.26 | kfife | I'll report the bug to Redhat. CENTOS will follow. We happen to have a RHEL5 support contract, so they'll at least listen to us. |
| 20:27.31 | Qwell | kfife: the bug report is only for the latter issue |
| 20:27.55 | mchou | am I missing something? There is not syntax like 'register => IAX2/user:password@provider/exten' to direct inbound IAX calls to a specific extension? |
| 20:28.07 | mchou | s/not/no |
| 20:29.35 | mchou | I mean you can do 'register => IAX2/user:password@provider' but not 'register => IAX2/user:password@provider/exten'? |
| 20:31.18 | [TK]D-Fender | mchou: You don't specify the protocol in its own driver file. thats redundant. |
| 20:31.23 | [TK]D-Fender | ok, heading home, later all |
| 20:31.58 | [gnubie] | thanks guys.. i have to go now.. i will just build asterisk and zaptel using gcc-4.1 and below |
| 20:31.58 | jbot | no worries, [gnubie] |
| 20:33.34 | jblack | I've never given the protocol in iax.conf for register. For troublesome providers, I make a special provider context in the dialplan, and goto them to the real inbound context. |
| 20:33.58 | [gnubie] | waves to all.. |
| 20:34.42 | *** join/#asterisk `paul (n=paul@125.252.68.126) |
| 20:34.57 | `paul | can i transfer a call using manager api? |
| 20:35.03 | jblack | I'm not sure where you got the idea that you can specify an extension with register. I can't find it. |
| 20:35.32 | jblack | `paul: I've never used AMI, but I think it does things like that. |
| 20:37.27 | kfife | jbot: Worked to de-select xpp. Question to apply the PATCH. I simply wget the patch "wget 'http://bugs.digium.com/file_download.php?file_id=19260&type=bug' -O - | patch -p0" and then recompile? |
| 20:37.37 | kfife | or do I need to make clean, and recompile? |
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| 20:38.13 | LemensTS | mchou: I will just do text to speech on the date and time. Everything else will be pre-recorded. I could write this myself, I just usually try to start with an open source project to save time, then modify it. I imagine ill be connecting to an accounting systems database for the date/time/phonenumber |
| 20:38.25 | amessina | is back. |
| 20:38.39 | LemensTS | mchou: the built in saynumber cmd will be fine for date/time |
| 20:38.51 | LemensTS | (or saydigit i forget what it was off top of my head) |
| 20:39.44 | LemensTS | Also figured there would be some AGI involved, im not good at that |
| 20:41.35 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 20:42.04 | blebleble | i may be mis understanding this but shouldnt 1513+NXXXXXX in a dialpattern automatically add 1513 to any 7 digital number dialed? |
| 20:42.07 | ghenry | why would crc_ccitt load up? which calls zaptel |
| 20:42.17 | ghenry | which doesn't source /etc/sysconfig/zaptel |
| 20:42.23 | ghenry | which has an extra module param in it |
| 20:44.11 | kfife | blebleble: no, but exten => NXXXXXX,1, Dial(1513${EXTEN}) will add 1513 to any 7 digit number dialed |
| 20:44.21 | kfife | provided the first digiti is not a zero or a one |
| 20:46.02 | kfife | blebleble: you obviously need to add your channel technology to the argumetn. I forgot to: Dial(local/1513${EXTEN}@locals) |
| 20:46.23 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
| 20:46.32 | kfife | blebleble: or add SIP/ etc |
| 20:48.48 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 20:49.20 | jblack | Hmmm. I'm getting different information on ToS based on where I look. |
| 20:49.45 | kfife | Thanks everyone. Zap has been successfully compiled! |
| 20:50.00 | kfife | using the patch, not using the workaround |
| 20:50.46 | jblack | The wiki, though, seems to imply I want 0x18, because the bits are shifted. |
| 20:51.28 | jblack | never mind. I can't count. |
| 20:52.44 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:53.45 | jblack | no, the wiki, and wikipedia are off by a bit. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf http://en.wikipedia.org/wiki/Type_of_Service |
| 20:54.30 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 20:54.45 | jblack | If wikipedia is right, 0x18 should really be 0x28 |
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| 20:57.04 | kfife | Anyone: What's the relationship between libpri and chan_misdn? Anyone use it? Is it for non-digium ISDN-PRI interfaces? |
| 20:57.41 | lowtek | Truley platonic but I hear libpri gets around ... |
| 20:58.02 | kfife | lowtek: :-) |
| 20:58.19 | kfife | be careful where you stick your RJ45 |
| 20:58.46 | lowtek | lol, ouch |
| 21:01.26 | jblack | heh. wikipedia is wrong too, on the priority |
| 21:02.46 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 21:05.08 | jblack | I think 0x18 should be replaced with 0x1a (Priority, Low delay, High Throughput) |
| 21:14.34 | jblack | Oh yeah. 0x1a is good. |
| 21:16.47 | x86 | ok, so I want to setup an intercom with Asterisk |
| 21:16.52 | x86 | paging, I guess you could call it |
| 21:17.24 | [TK]D-Fender | x86: We have reached a concensus and you have our permission. Go for it. |
| 21:17.33 | x86 | is there something I can plug into an analog port that would auto-answer, have a built-in amplifier, and speaker? |
| 21:17.47 | x86 | [TK]D-Fender: let me finish there sparky :) |
| 21:17.56 | [TK]D-Fender | x86: Lookup the Viking paging usints from the WIKI |
| 21:18.09 | [TK]D-Fender | unist* |
| 21:18.15 | [TK]D-Fender | units* |
| 21:18.17 | [TK]D-Fender | damnit |
| 21:19.03 | x86 | lol |
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| 21:20.24 | x86 | thanks |
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| 21:21.15 | *** join/#asterisk itakinet7 (n=chatzill@adsl-065-005-186-231.sip.asm.bellsouth.net) |
| 21:21.48 | itakinet7 | AAUG Meeting - Tuesday, August 26th at 7PM EDT Details http://atlaug.com/blog/ |
| 21:23.50 | kfife | Anybody here in Chicago? I'm looking for a Chicago Asterisk user's group. |
| 21:24.09 | x86 | [TK]D-Fender: got a specific model? |
| 21:24.15 | jblack | CAUG sounds close to a cool name. |
| 21:24.17 | itakinet7 | http://www.asterisk.org/community |
| 21:24.47 | kfife | jblack: thanlks |
| 21:24.51 | kfife | jblack: thanks |
| 21:25.09 | jblack | anyways, check meetup.com |
| 21:25.51 | [TK]D-Fender | x86: Nope. |
| 21:27.06 | amessina | kfife, i am. there is a chicago area group, but it's very non-active as far as i can tell. http://groups.google.com/group/asterisk-chicago |
| 21:27.06 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583027.dsl.bell.ca) |
| 21:28.51 | kfife | amessina: I looked into the user's group a while back but I couldn't get a reply from the coordinator. Perhaps we can kick-start it. |
| 21:29.19 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:29.49 | amessina | kfife. the same with me. i joined. it took about 8 months for approval. they had one talk and eat pizza meeting that i couldn't make it to. i'm not sure how much they accomplished. |
| 21:30.25 | x86 | [TK]D-Fender: I called viking, they've got this awesome little CPA-7B... auto-answers an analog extension and has built-in amplifier and comes with 1 loudspeaker |
| 21:31.03 | [TK]D-Fender | x86: Cool. |
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| 21:58.25 | tvirus | Has anyone been able to setup HUDLite on a non-trixbox machine? I'm just getting it set up and trying to call from softphone to softphone inside of the office. Asterisk says 'cannot create channel SIP/user'. HUDLite tosses out some random error. |
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| 22:20.46 | jblack | where is everyone today? |
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| 22:28.17 | jameswf-home | ~centos52bug |
| 22:28.17 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
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| 22:43.44 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 22:44.14 | mltlnx | How can I tell what a version of zaptel i am running? |
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| 22:45.43 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 22:45.50 | pcrane | hi guys |
| 22:46.39 | pcrane | I'm trying to work out where the 'name' message is played in voicemail |
| 22:46.46 | pcrane | there's a setting for recording it... |
| 22:46.52 | pcrane | but under what conditions is it played back? |
| 22:47.28 | riddlebox | usually name is played when you are in a directory I believe |
| 22:48.05 | outtolunc | mltlnx: dmesg |grep "Zaptel Version:" |
| 22:48.29 | pcrane | it's in the same category as recording busy and unavailable messages... |
| 22:48.54 | mltlnx | thanks |
| 22:51.25 | pcrane | hmm... |
| 22:51.31 | pcrane | maybe it's to do with the directory |
| 22:57.34 | outtolunc | pcrane: a grep play_mailbox_owner * |
| 22:57.49 | pcrane | where abouts? |
| 22:57.53 | outtolunc | in the main dirs shows it in app_directory.c |
| 22:58.45 | pcrane | ok |
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| 23:00.46 | pcrane | cheers outtolunc |
| 23:00.59 | outtolunc | np |
| 23:01.11 | pcrane | half the problem is knowing where to look for the information |
| 23:01.27 | pcrane | has gotten good at looking at the wiki... |
| 23:01.41 | pcrane | needs to remember about the source code and show application <something> |
| 23:01.43 | pcrane | ;) |
| 23:02.08 | outtolunc | as fast as things change around here, i always look to the source code |
| 23:02.09 | heedly | the source is kinda difficult. |
| 23:02.26 | pcrane | at least it's nice to look at |
| 23:02.30 | pcrane | some of the code I see is horrible |
| 23:02.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:05.50 | jameswf-home | ~centos52bug |
| 23:05.50 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 23:06.12 | knarfly | ~book |
| 23:06.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 23:54.01 | HumanCell | Woohoo! Found the right iptables syntax to open the correct port-range ... |
| 23:54.07 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:58.37 | anonymouz666 | why after you set the iax2 peers in iax.conf just removing them from the file and 'iax2 reload' does not remove the peers from 'iax2 show peers', why? just if i do "restart now" |