| 00:01.13 | cvnet | i have to install a vpn client correct? |
| 00:06.37 | ricko73 | cvnet: racoon does ipsec |
| 00:06.47 | ricko73 | google has information on racoon |
| 00:06.52 | ricko73 | coincidence? |
| 00:07.02 | ricko73 | ~google |
| 00:07.03 | infobot | from memory, google is http://lmgtfy.com/?q=google |
| 00:09.39 | Qwell | ~cioncidence |
| 00:09.42 | Qwell | ~coincidence |
| 00:09.43 | Qwell | .. |
| 00:11.31 | *** join/#asterisk l2trace99 (n=jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 00:16.14 | cvnet | i have installed ipsec-tools and racoon, now do i change the racoon.conf or leave it as is? |
| 00:19.46 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:26.17 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 00:27.48 | [T]ank | does anyone else ever have the issue of a 10-15 second delay before your outbound call is really placed? I have this on a few different asterisk servers. Some are on T1, SIP or IAX2. Not sure what it is. The call ALWAYS is successful, but there is always a delay before the call is actually placed. Wondering if anyone else has had that issue or knows how to correct it. |
| 00:27.54 | [T]ank | just a general annoyance. |
| 00:30.44 | [TK]D-Fender | [T]ank: Yes you are, but we forgive you ;) |
| 00:31.01 | [T]ank | lol... sorry. |
| 00:31.23 | [T]ank | ever seen that before? |
| 00:31.37 | [TK]D-Fender | [T]ank: Now try DETAILS. Show us debug, tell us what version. Describe the networking, etc |
| 00:31.52 | [TK]D-Fender | ~wmmfpb |
| 00:31.52 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 00:31.54 | [TK]D-Fender | :p |
| 00:32.07 | [T]ank | gathering.... |
| 00:32.17 | Qwell | [TK]D-Fender: it's on the plane with the snakes. |
| 00:32.37 | [TK]D-Fender | fetches Samuel L. Jackson .... FOR GREAT JUSTICE! |
| 00:34.12 | jaytee | hahahaha |
| 00:34.42 | jaytee | you people are sick! I knew there was a reason I felt at home here :-) |
| 00:34.42 | *** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30) |
| 00:35.32 | [TK]D-Fender | jaytee: I'm not sick... but I'm not well ;) |
| 00:38.49 | justdave | I'm trying to set up a T1/PRI and still not used to the dahdi stuff, is there a way I can tell if the link is up on the T1 line from the shell on the machine? |
| 00:39.07 | justdave | (this is my first time touching a PRI before, too) |
| 00:39.08 | SaiSoma|AFK | try service dahdi status |
| 00:39.29 | justdave | showing RED all the way down |
| 00:39.39 | SaiSoma|AFK | i would guess that it's not up then |
| 00:39.40 | justdave | which I already knew from the asterisk log. :) trying to figure out why |
| 00:40.06 | SaiSoma|AFK | i haven't setup a PRI with asterisk yet tho. i use the 8 port analog and red means not functional |
| 00:40.37 | SaiSoma|AFK | this is the only dahdi card in the box? |
| 00:40.45 | justdave | yep |
| 00:41.04 | SaiSoma|AFK | mmmm . is it the local carrier on the other end i guess? |
| 00:41.40 | SaiSoma|AFK | meaning you aren't connecting to another phone system of your own? |
| 00:41.54 | justdave | I'm on the phone with them, they're saying there's no link :| |
| 00:41.59 | justdave | it's between us and the phone company |
| 00:42.52 | SaiSoma|AFK | *nod* so, can they loop the smartjack at your location? |
| 00:43.27 | justdave | aha, found it. |
| 00:43.33 | SaiSoma|AFK | awesome, what was it? |
| 00:43.39 | justdave | my network guy says they punched it down in the wrong port |
| 00:43.43 | SaiSoma|AFK | (i ask because i'll be doing this in about two weeks, heh) |
| 00:43.46 | SaiSoma|AFK | ahhh, awesome |
| 00:43.46 | justdave | he just swapped it over, alarm went away |
| 00:43.53 | SaiSoma|AFK | excellent. good luck! |
| 00:44.37 | [TK]D-Fender | Off for a while, BBL |
| 00:45.57 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
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| 01:31.11 | paulius | Okay so people will kill me here for this but... |
| 01:31.21 | paulius | Is there anyone here familiar with Cisco IP phones with SIP firmware? |
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| 01:39.03 | ke5eju | is anyone in here an app_rpt guru? |
| 01:39.26 | Qwell | ke5eju: it would surprise me very little if nobody in here had ever used it. |
| 01:40.03 | ke5eju | wow... well.. ok where would i go if i'm having trouble tuning the radio with app rpt |
| 01:40.35 | Qwell | I'm sure there are people who use it, but I don't know where they are |
| 01:40.56 | ke5eju | okay thanks |
| 01:45.02 | jks | it's possible to pause/unpause an interface in regards to queues using the manager interface - is there any way to read out the current pause status for an interface? |
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| 01:50.05 | Qwell | jks: there should be a PauseQueueMember (and one for unpause..) event |
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| 01:51.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:54.08 | johnakabean | hey everyone; I had a problem with centos and asterisk was automatically restarting every 10 or 15 minutes; it turned out to be bad memory. I have put the hard drives in another, much newer, computer (to my surprise I didn't even have to recompile centos). Asterisk is still having the same problem but It ONLY restarts automatically a few minutes after using freepbx to reload it |
| 01:54.44 | johnakabean | The queues have stopped working; If I make freepbx reload asterisk, the queues will work again for a few minutes until asterisk automatically restarts |
| 01:55.01 | johnakabean | I have reinstalled freepbx, recompiled asterisk (after make clean and ./configure) and its addons. |
| 01:57.43 | johnakabean | any suggestions? |
| 02:06.04 | jks | Qwell, hmm, yes but I'm trying to just do a "check" to see the status |
| 02:06.36 | jks | Qwell, i.e. in the case the client hasn't been connected to the manager interface before, it then connects... and I want to be able to tell if the interfaces are paused or unpaused |
| 02:07.31 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 02:08.05 | Qwell | there should be a way to get the status... have you looked at the list of manager events? |
| 02:08.24 | jks | Qwell, yes, I don't see how events will help me? |
| 02:08.37 | jks | Qwell, I mean, it should be some kind of command that will list the statuses I guess? |
| 02:08.37 | *** join/#asterisk johnakabean (n=john@static-173-50-101-10.nrflva.east.verizon.net) |
| 02:08.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:08.41 | Qwell | actions, sorry |
| 02:09.03 | jks | I have looked at them yes, but haven't found anything that worked |
| 02:09.10 | johnakabean | qwell, you know how to wipe asterisk and freepbx completely (minus config /etc/asterisk)? |
| 02:09.55 | jks | Qwell, I can use the Command to do a queue show command and then checked for (paused)... but it will only display pause status for the individual queues - I cannot tell if the interface as such is paused |
| 02:10.04 | *** join/#asterisk aliverius (n=aliveriu@chal530-a119.home.otenet.gr) |
| 02:12.46 | johnakabean | [Jun 5 18:58:36] VERBOSE[20633] logger.c: == Manager unregistered action QueueStatus |
| 02:12.46 | johnakabean | [Jun 5 18:58:36] VERBOSE[20633] logger.c: == Manager unregistered action Queues |
| 02:12.46 | johnakabean | [Jun 5 18:58:36] VERBOSE[20633] logger.c: == Manager unregistered action QueueAdd |
| 02:12.46 | johnakabean | [Jun 5 18:58:36] VERBOSE[20633] logger.c: == Manager unregistered action QueueRemove |
| 02:12.46 | johnakabean | [Jun 5 18:58:36] VERBOSE[20633] logger.c: == Manager unregistered action QueuePause |
| 02:12.49 | johnakabean | what is that? |
| 02:12.52 | johnakabean | sorry for flood |
| 02:13.13 | jks | johnakabean, what is what? |
| 02:13.20 | johnakabean | Manager unregistered action Queue |
| 02:13.31 | johnakabean | my queues aren't working |
| 02:13.43 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 02:14.17 | drmessano | Asterisk 1.6? |
| 02:15.32 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 02:15.41 | johnakabean | 1.4 DR messano |
| 02:16.10 | johnakabean | DR messano, I figured out what was causing that segmentation fault |
| 02:16.12 | johnakabean | BAD MEMORY |
| 02:16.51 | johnakabean | I'm running the same install/setup on a completely different, newer machine |
| 02:16.56 | drmessano | I bet that got rid of those segmentation fault messages too |
| 02:17.06 | johnakabean | asterisk is still restarting itself but NOT as frequently |
| 02:17.19 | johnakabean | it only does it ONCE after I use freepbx to reload it |
| 02:17.56 | johnakabean | what makes me use freepbx to reload it is so my queues will prevail to work for only 10 minutes |
| 02:18.01 | johnakabean | then asterisk restarts itself |
| 02:18.37 | johnakabean | I am following a forum suggestion to start asterisk in the console |
| 02:18.38 | drmessano | Are you running zaptel/dahdi? |
| 02:18.41 | johnakabean | zaptel |
| 02:18.48 | drmessano | Did you recompile it? |
| 02:18.55 | *** join/#asterisk sehro (n=rautenkr@97-122-128-212.hlrn.qwest.net) |
| 02:19.04 | johnakabean | anyway, DR, I'm starting asterisk in the console so that when it crashes it will create a dumb |
| 02:19.06 | johnakabean | dump |
| 02:19.12 | drmessano | Did you recompile it? |
| 02:19.15 | drmessano | zaptel |
| 02:19.28 | Qwell | ~drmessano |
| 02:19.29 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 02:20.34 | drmessano | exactly |
| 02:21.11 | johnakabean | I will try that |
| 02:21.24 | johnakabean | I did recompile asterisk and reinstalled freepbx |
| 02:21.41 | johnakabean | my next thing was to check the mysql tables for corruption |
| 02:21.48 | drmessano | Zaptel I would think would be sensitive to moving to different hardware |
| 02:22.03 | johnakabean | I don't use any hardware other than the dummy |
| 02:22.16 | johnakabean | I'm all SIP/IAX2 |
| 02:22.26 | drmessano | You have a CPU and a motherboard, dont you? |
| 02:22.43 | johnakabean | ok whatever recompiling |
| 02:22.56 | johnakabean | would you recommend dahdi over zaptel |
| 02:22.59 | johnakabean | ? |
| 02:23.04 | drmessano | Yes |
| 02:23.13 | drmessano | Zaptel is old and busted |
| 02:23.19 | drmessano | Dahdi is the new hotness |
| 02:23.25 | johnakabean | do i just compile it instead of zaptel without having to change configs for asterisk? |
| 02:23.47 | johnakabean | of course I recompile zaptel just to do a make uninstall |
| 02:23.59 | drmessano | No, use Dahdi to uninstall zaptel |
| 02:24.14 | drmessano | Dahdi installer removes zaptel where zaptel doesnt remove itself |
| 02:24.14 | johnakabean | it will give me an option or does it just overwrite it? |
| 02:24.19 | johnakabean | ok |
| 02:24.32 | drmessano | Not sure.. I did mine the hard way, and found out later I wasted time |
| 02:25.11 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 02:25.15 | johnakabean | I'm tempted to recompile ALL of centos in 64 bit |
| 02:25.19 | johnakabean | but this pc is a loaner |
| 02:25.21 | johnakabean | :) |
| 02:25.57 | johnakabean | usually takes 5 minutes on my machine to just compile asterisk |
| 02:26.01 | johnakabean | this machine took 20 seconds |
| 02:26.06 | johnakabean | in 32 bit |
| 02:26.23 | drmessano | nice |
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| 02:26.58 | *** part/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 02:27.07 | johnakabean | do I need dahdi tools? |
| 02:27.15 | johnakabean | since I have no hardware |
| 02:28.15 | drmessano | I just get the whole package.. |
| 02:28.47 | johnakabean | well, there are two listed; one is dahdi and the other is tools for it |
| 02:28.48 | drmessano | But probably not |
| 02:29.14 | drmessano | I download the dahdi+tools |
| 02:30.04 | drmessano | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0-rc5+2.2.0-rc3.tar.gz |
| 02:30.15 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 02:30.19 | drmessano | Then I have it if I need it later |
| 02:32.17 | aliverius | my distro was comming with a very bloated buildscript |
| 02:32.33 | johnakabean | No hardware timing source found in /proc/dahdi, loading dahdi_dummy |
| 02:32.39 | johnakabean | that looks familiar |
| 02:32.43 | aliverius | i guess it is for a full server but it took me much time to realise what i need and what not |
| 02:33.26 | johnakabean | ok queues still not working |
| 02:36.04 | johnakabean | Jun 5 22:34:44] WARNING[11481] acl.c: all is not a valid IP |
| 02:36.20 | johnakabean | thats in the console |
| 02:36.33 | johnakabean | how do I set the ips to listen on? |
| 02:40.21 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 02:49.06 | *** part/#asterisk aliverius (n=aliveriu@chal530-a119.home.otenet.gr) |
| 02:50.51 | luckyaba | anyone have a recommendation on a firewall appliance that does a good job with QOS? |
| 02:53.55 | johnakabean | your own centos |
| 02:53.59 | johnakabean | or linux |
| 02:54.28 | johnakabean | lucky, go here |
| 02:54.29 | luckyaba | Well I would rather not have the box wide open to the web |
| 02:54.42 | luckyaba | and the Sonicwall in place at the moment does a poor job at QOS |
| 02:54.54 | johnakabean | if you do iptables right, it will be the best firewall you have ever had |
| 02:55.02 | johnakabean | you use iptables, snort, and ctshaper |
| 02:55.52 | johnakabean | I have 733 rules in my iptables |
| 02:56.00 | johnakabean | it is STRICT |
| 02:56.18 | johnakabean | the chinese have been trying to hack me for months |
| 02:56.20 | johnakabean | lol |
| 02:56.37 | jaytee | the chinese suck |
| 02:56.45 | luckyaba | what if any are you doing for intrusion prevention/detection |
| 02:56.48 | johnakabean | i was giving an example |
| 02:56.55 | johnakabean | fail2ban and snort |
| 02:57.07 | johnakabean | fail2ban takes care of brute forcwe |
| 02:57.12 | johnakabean | snort does the other |
| 02:57.27 | johnakabean | iptables also takes care of syn flood and other things |
| 02:57.32 | luckyaba | ok, given that sounds like a pretty locked down box |
| 02:57.38 | luckyaba | how much setup are we looking at :P |
| 02:57.50 | johnakabean | go install snort |
| 02:57.56 | johnakabean | that will take the longest |
| 02:58.09 | luckyaba | I am looking for a solution for not only our network but for clients as well |
| 02:58.21 | johnakabean | ahh but I have a network of 12 |
| 02:58.37 | johnakabean | behind this one box |
| 02:58.49 | smps | luckyaba, depens how much money you would like to invest |
| 02:59.04 | smps | luckyaba, imho i would use linux for stuff in my company |
| 02:59.21 | smps | luckyaba, for clients i use juniper and cisco |
| 02:59.34 | luckyaba | no, I mean we are going to be deploying VOIP servers in the coming months to clients who are almost all running Sonicwalls and Windows |
| 02:59.59 | johnakabean | good luck with SIP on sonic |
| 03:00.58 | johnakabean | I use a utility that helps me with iptables |
| 03:01.07 | johnakabean | i will help you with that after you install snort |
| 03:01.30 | johnakabean | i type about 50 lines and it generates a sound firewall for iptables |
| 03:02.03 | johnakabean | about 733 from me ranging to internal network servers to quality of service for voip to securing the box itself |
| 03:02.33 | johnakabean | if you don't want problems with nat traversal, you need to have the asterisk box public |
| 03:02.47 | johnakabean | ONLY my interal clients can connect to the asterisk box |
| 03:03.01 | johnakabean | asterisk is allowed to accessing the trunks |
| 03:03.05 | luckyaba | Well there are appliances that deal with that problem which is what I was leaning towards since its a solution we are providing to clients |
| 03:03.06 | johnakabean | and they are limited by ip |
| 03:03.29 | johnakabean | asterisk can only communicate with my trunks |
| 03:03.35 | johnakabean | and is open to the internal network |
| 03:03.36 | luckyaba | I am all for linux and if it was my network I would totally do it |
| 03:03.55 | johnakabean | well, I invite you to test my network setup |
| 03:04.10 | johnakabean | 173.50.101.10 - 15 |
| 03:04.12 | luckyaba | It took me almost a year to get them to accept an open source solution for VOIP installs! |
| 03:04.13 | luckyaba | lol |
| 03:04.52 | johnakabean | you can't spoof your mac to me or your ip address......a bunch of things this neat script secures |
| 03:05.52 | luckyaba | I will setup a test lab and check it out though. If I can script most of it and get the install fairly automated and quick I could probably get the big guys on board |
| 03:07.14 | johnakabean | so dahdi did fix that |
| 03:07.45 | luckyaba | You have any alerting setup on your network? |
| 03:07.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:10.52 | smps | johnakabean, you can use sip nat helper for iptables to address nat traversal issues |
| 03:10.59 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
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| 04:58.31 | AlmightyOatmeal | i'm getting really poor quality music on hold and i get this in the console: [Jun 5 23:54:39] NOTICE[65484]: rtp.c:1280 ast_rtp_read: Unknown RTP codec 126 received from '...' |
| 04:58.40 | AlmightyOatmeal | does that mean there is a codec issue with MOH? |
| 05:04.47 | *** join/#asterisk DarthPointer (n=no@216.68.218.82) |
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| 05:17.01 | dpreacher | is it possible to check with the asterisk command when the last iax2 reload was performed. I'm monitoring IP changes and I'm keeping a log of IP changes and also issuing a reload. Although I know when the reload is being triggered I have no way to check if actually the reload took place. is there a return code or a log that I can check to see if the reload had failed due to any circumstances? |
| 05:17.09 | dpreacher | thank you in advance |
| 05:19.06 | dpreacher | if people aren't responding in this room, can i ask in a related channel or will there be useless blah blah about cross posting even if no one helps me here |
| 05:19.33 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 05:27.57 | AlmightyOatmeal | google? |
| 05:28.17 | AlmightyOatmeal | or maybe no one is available currently and someone will get back to you once they 1) are available 2) know an answer |
| 05:30.03 | dpreacher | AlmightyOatmeal, can i ask in asterisknow...coz i've seen some people unnecessarily using foul language, just coz they see the question posted in another channel they are in |
| 05:30.49 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
| 05:31.51 | AlmightyOatmeal | dpreacher: whatever floats your boat |
| 05:32.13 | dpreacher | ok i asked there |
| 05:32.22 | dpreacher | also...trying google |
| 05:35.02 | *** part/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 05:39.57 | dpreacher | AlmightyOatmeal, do you use any certain Asterisk specific distribution? |
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| 06:07.20 | dpreacher | please note sfire on asterisknow has helped me in asterisknow, so thanks for looking at the question and dreaming |
| 06:10.28 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
| 06:16.46 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 06:22.15 | *** join/#asterisk rajmohan (n=rajmohan@122.165.25.171) |
| 06:23.37 | rajmohan | hi, iam getting this error when i use asterisk -r, Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) any help pls? |
| 06:25.18 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
| 07:12.55 | AlmightyOatmeal | rajmohan: your current user doesn't have access to asterisk.ctl, if your asterisk is running as root, then su or sudo and run asterisk -r |
| 07:14.54 | rajmohan | Almighty: now i got is working it seems it says Connected to Asterisk 1.4.25 currently running on ubuntu |
| 07:15.18 | rajmohan | can you guide me now how can i open up the amportal? |
| 07:15.26 | rajmohan | it gives 404 error |
| 07:15.42 | rajmohan | but the folder is there and iam not able to view the files in it |
| 07:28.18 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 07:41.05 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 07:54.18 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:04.28 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 08:06.19 | iksik | morning |
| 08:06.29 | iksik | is there any nice opensource billing system for asterisk? |
| 08:13.44 | *** join/#asterisk rethus (n=rethus@81.173.255.83) |
| 08:13.53 | rethus | Hi There, |
| 08:14.14 | rethus | i have * 1.4 in use. what is the difference to 1.6.... many things are changed? |
| 08:23.46 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:27.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-88ec84683955e313) |
| 08:29.16 | *** join/#asterisk stijnbe (n=stijnbe@84.193.98.70) |
| 08:35.25 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 08:35.46 | rethus | if i start asterisk 1.4 my cpu usage jumpo to 100% whats going on... how can i check what the problem is? |
| 08:42.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:53.41 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:55.26 | rethus | hello... nobody active? 258 Persons and no traffic here? Or is my messenger broken? |
| 09:09.58 | *** join/#asterisk jgoo (n=r3rman@adsl141-37.lsf.forthnet.gr) |
| 09:12.40 | jgoo | hrm - what the hell. So I have an asterisk box and 20 grandstream 201s - set them up, all registering - I move them, reconnect them, they are all online, I can ping all devices from the pbx, but none are registering - what can cause that? |
| 09:16.54 | rethus | does asterisk 1.4 has no dial() function on cmd anymore? |
| 09:35.11 | iksik | hm |
| 09:35.31 | iksik | asterisk is locking CDR csv files while running? |
| 10:28.24 | *** join/#asterisk BlackSlik (n=james@41.219.248.208) |
| 10:40.52 | BlackSlik | login as: root |
| 10:40.52 | BlackSlik | root@76.73.56.50's password: |
| 10:40.52 | BlackSlik | Last login: Sat Jun 6 11:35:48 2009 from 41.219.248.208 |
| 10:40.52 | BlackSlik | [root@nextdreamnet ~]# cd /usr/src/ |
| 10:40.52 | BlackSlik | [root@nextdreamnet src]# dir |
| 10:40.53 | BlackSlik | asterisk-1.4.26-rc1 dahdi-linux-2.2.0-rc5 |
| 10:44.52 | *** join/#asterisk BlackSlik (n=james@41.219.248.208) |
| 10:45.38 | *** part/#asterisk rethus (n=rethus@81.173.255.83) |
| 10:52.18 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 10:59.57 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.95) |
| 11:02.26 | *** join/#asterisk puzzled (n=foobar@blahfoo.xs4all.nl) |
| 11:12.07 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 11:12.36 | *** join/#asterisk aliverius (n=aliveriu@chal530-a019.home.otenet.gr) |
| 11:19.11 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 11:27.27 | *** join/#asterisk jmacz (n=jmacz@190.25.38.42) |
| 12:06.12 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 12:13.41 | *** join/#asterisk chainsawbike (n=chainsaw@121-73-114-254.cable.telstraclear.net) |
| 12:15.58 | *** join/#asterisk chainsawf (n=chainsaw@121-73-114-254.cable.telstraclear.net) |
| 12:22.57 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:22.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:32.50 | *** join/#asterisk chendy (n=alex@121.35.145.24) |
| 12:43.45 | *** join/#asterisk gazzerh (n=garryh@93-97-187-150.zone5.bethere.co.uk) |
| 12:47.40 | *** join/#asterisk bmg505 (n=leon@196-209-79-120-rndf-esr-5.dynamic.isadsl.co.za) |
| 12:57.23 | *** join/#asterisk DarkRift (n=dark@65.92.165.162) |
| 13:02.57 | *** join/#asterisk smace (n=smace@187.16.246.3) |
| 13:04.59 | smace | hello. Could anybody suggest me one free sip client for windows that has no problems with queue? Currently i'm using Zoiper but it rins in all phones when a new call is received. |
| 13:05.33 | smace | *rings |
| 13:06.33 | *** join/#asterisk crazyhick1 (n=carrie@belssmcnas07-3521295200.dial.bell.ca) |
| 13:11.34 | *** join/#asterisk PseudoNim (n=pseudo@modemcable163.0-22-96.mc.videotron.ca) |
| 13:11.54 | PseudoNim | hi all |
| 13:12.33 | PseudoNim | in my dialplan, i have this string: exten => _011.,1,Dial(sip/voipprovider/${EXTEN}). i would like, however, to strip "011" from the numbers sent by my client ........ because my client sends it twice (011011xxxxxx). how can i just strip the first 4 digits say? |
| 13:12.45 | PseudoNim | do i just say 4|_011? |
| 13:13.01 | *** join/#asterisk aliverius (n=aliveriu@chal530-a019.home.otenet.gr) |
| 13:14.53 | smps | PseudoNim, in that case it will strip number 4 from digits |
| 13:15.13 | PseudoNim | oops heh |
| 13:15.14 | smps | PseudoNim, as if you were dialing 401234 it will strip 4 and dial 01234 |
| 13:17.21 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 13:17.40 | PseudoNim | hmm, changing it to 011|_011.,1,.......etc didn't work, gave an 'extension not found' |
| 13:17.59 | smps | hehe |
| 13:18.08 | PseudoNim | sorry, noob |
| 13:18.12 | PseudoNim | hehe |
| 13:19.09 | barbacha | PseudoNim: did you try exten => _011.,1,Dial(sip/voipprovider/${EXTEN:3}) ? |
| 13:19.21 | PseudoNim | ooh, nope |
| 13:19.22 | PseudoNim | let me try |
| 13:22.07 | smace | I need one good SIP client, any suggestion? |
| 13:22.12 | smps | smace, x-lite |
| 13:23.42 | *** join/#asterisk dnikulin (n=den@ws12.amber.pu.ru) |
| 13:23.45 | dnikulin | hi |
| 13:23.46 | barbacha | smps: twinkle under linux |
| 13:24.11 | smps | barbacha, he was asking for windows sip client |
| 13:24.31 | dnikulin | how can I make work two clients from 1 IP address without error "check_auth: username mismatch, have <111>, digest has <222> |
| 13:24.31 | dnikulin | " |
| 13:26.07 | aliverius | hi! i am new to asterisk and i downloaded the oreilly book. would it help me with asterisk 1.6? |
| 13:26.20 | smace | when using x-lite with ilbc the voices sounds like robots. this is the why i was using zoiper instead. but zoiper rings when there is a new call in queue. |
| 13:27.35 | drmessano | iLbc is horrid though |
| 13:28.02 | drmessano | Way too much CPU overhead for the gain |
| 13:28.21 | drmessano | Maybe its better than it sounds so bad :) |
| 13:30.21 | *** join/#asterisk _ShrikE (n=_ShrikE@74.185.215.60) |
| 13:32.50 | smace | drmessano: which codec do you suggest? |
| 13:36.17 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 13:37.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:42.29 | PseudoNim | barbacha: nice, that worked. thanks! |
| 13:43.25 | PseudoNim | a harder question though... how can i do string manipulation? |
| 13:43.39 | PseudoNim | basically... the standalone phone i'm using sends EVERYTHING as 011xxxxxxxxx |
| 13:44.02 | PseudoNim | oh .... wait |
| 13:44.04 | PseudoNim | i think i know. |
| 13:44.17 | PseudoNim | i just need to fix the rule that says _1NXXNXXXXXX |
| 13:46.24 | PseudoNim | yep, that worked. cool |
| 13:53.17 | *** join/#asterisk rethus (n=rethus@81.173.255.83) |
| 13:54.08 | rethus | anyone know how can i use a dial() into cli via "manager show commands Command"? |
| 13:54.46 | rethus | before i used phpagi with: $as->Command('dial 100@phones'); and it workt, but for now it doesn't work anymore... i don't know why |
| 14:00.41 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:05.04 | *** part/#asterisk rethus (n=rethus@81.173.255.83) |
| 14:09.43 | iksik | hum |
| 14:09.52 | iksik | asterisk should listen on 1720 port for h323 ? |
| 14:13.56 | kaldemar | iksik: it listens to the port you tell it to. |
| 14:14.07 | iksik | yeah I know... |
| 14:14.12 | iksik | but it doesn't listen :P |
| 14:14.26 | kaldemar | do you have the module loaded? |
| 14:14.26 | iksik | oh damn ;/ |
| 14:14.29 | iksik | chan_h323.so: cannot open shared object file |
| 14:14.40 | iksik | no such file or directory |
| 14:14.45 | iksik | hm |
| 14:21.18 | *** part/#asterisk dnikulin (n=den@ws12.amber.pu.ru) |
| 14:36.15 | *** join/#asterisk propellerhead (n=yogurt2u@host237.190-31-75.telecom.net.ar) |
| 14:36.57 | aliverius | does chan_misdn require libpri? |
| 14:37.23 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 14:51.24 | *** join/#asterisk spiri (n=spiri@S01060002553240a8.vc.shawcable.net) |
| 14:54.58 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 14:55.30 | spiri | Hi, can I put a forieng sip extension in a call file? the examples I have seen seem to be local |
| 15:02.24 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 15:03.23 | jaytee | morning [TK]D-Fender |
| 15:03.30 | [TK]D-Fender | *yawn* |
| 15:04.16 | jaytee | hands [TK]D-Fender a cup of fresh brewed Columbian "half & half, sugar?" |
| 15:05.18 | [TK]D-Fender | jaytee: That works... |
| 15:05.28 | [TK]D-Fender | gonna go boil a pot for it now... |
| 15:05.33 | jaytee | [TK]D-Fender, late night? |
| 15:06.01 | [TK]D-Fender | jaytee: Not so much, but slept funny. |
| 15:06.10 | drmessano | He feels numb.. maybe something with lemon.. shouldn't taste too bad |
| 15:06.18 | drmessano | Yay.. U2 triple play |
| 15:06.44 | jaytee | puts on "Where the streets have no name" |
| 15:06.53 | jaytee | morning drmessano |
| 15:07.06 | drmessano | I just finished listening to Numb, Lemon, and now Bad.. and that worked out well |
| 15:07.08 | drmessano | Morning |
| 15:08.00 | jaytee | hahah, anyone been to Google today? Tetris FTW! |
| 15:08.24 | drmessano | That was cool |
| 15:12.31 | *** join/#asterisk Aiatek (n=munoz@190.159.121.197) |
| 15:13.17 | spiri | hi, can someone recomend some docs.. Im trying to place outbound sip calls with asterisk.. but they are failing.. I can easily place these calls with ekiga I am wondering what I am missing |
| 15:14.35 | drmessano | ~book |
| 15:14.36 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:18.27 | *** join/#asterisk timeshell (n=timeshel@76.70.209.246) |
| 15:20.45 | spiri | thanks.. drmessano I have this book in front of me |
| 15:21.52 | spiri | still cant place out going sip calls w/ a call file from asterisk |
| 15:23.49 | [TK]D-Fender | spiri: And you aren't showing us your failed attempt with SIP DEBUG from CLI |
| 15:23.53 | [TK]D-Fender | ~pb |
| 15:23.54 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 15:23.55 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:27.38 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 15:28.01 | *** join/#asterisk chendy (n=chatzill@121.35.145.24) |
| 15:28.10 | spiri | [TK]D-Fender: this is my call file http://pastebin.ca/1449775 |
| 15:28.36 | [TK]D-Fender | spiri: Extension: SIP/77223@voip.telephreak.org <- not valid |
| 15:28.49 | [TK]D-Fender | spiri: You seem to have understood this process backwards |
| 15:28.53 | spiri | this is the output |
| 15:28.55 | spiri | http://pastebin.ca/1449778 |
| 15:29.04 | spiri | oh.. likely |
| 15:29.08 | chendy | does anybody working on porting asterisk to opensolaris ? |
| 15:29.12 | [TK]D-Fender | spiri: "Extension" is the extension in your dialplan that the "Channel" will go to one answered |
| 15:29.25 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 15:30.29 | mltlnx | Is there any possibility of retrieving the email address that is in Voicemail.conf through the manager interface? |
| 15:31.28 | *** part/#asterisk PseudoNim (n=pseudo@modemcable163.0-22-96.mc.videotron.ca) |
| 15:31.57 | spiri | [TK]D-Fender: ahhh cool it seems to have connected |
| 15:32.03 | spiri | thanks |
| 15:47.12 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 15:47.16 | leifmadsen | For anyone who is interested in the new Asterisk release process, and would like more information about how it works, I've just written a post about it: http://leifmadsen.wordpress.com/2009/06/06/about-the-new-asterisk-versioning-method/ |
| 15:47.36 | jaytee | cool! thanks, leif |
| 15:47.42 | leifmadsen | I'd appreciate any feedback you can give, and let me know if any parts are confusing |
| 15:48.12 | leifmadsen | I'll try to address your comments as best I can -- leave the comments on the blog if you can so my site looks busy, lol |
| 15:48.23 | jaytee | will do |
| 15:48.57 | leifmadsen | merci! |
| 15:53.35 | kaldemar | leifmadsen: "Within the 1.6.1 branch, we'll have tags just like all our other branches; 1.6.1.0, 1.6.0.1, 1.6.0.2, etc." |
| 15:54.04 | leifmadsen | kaldemar: aha, thank you |
| 15:54.49 | drmessano | <PROTECTED> |
| 15:55.02 | leifmadsen | drmessano: yes |
| 15:55.16 | seanbright | leave leif ALONE!!! |
| 15:55.21 | leifmadsen | (I didn't proof read this article as I'm in a rush to get out of here, and down to the brewery tour :)) |
| 15:55.24 | seanbright | he's doing gawd's work |
| 15:55.34 | drmessano | Im trying to help him not sound like a douche |
| 15:55.39 | drmessano | Sorry :( |
| 15:55.49 | seanbright | drmessano: that's a full time job |
| 15:55.49 | jaytee | it's excellent |
| 15:55.57 | seanbright | leifmadsen <3 |
| 15:56.08 | drmessano | Otherwise, thats a great post |
| 15:56.15 | leifmadsen | drmessano: thanks! changes made |
| 15:57.16 | leifmadsen | thanks everyone! |
| 15:57.19 | leifmadsen | much appreciated |
| 15:57.23 | leifmadsen | I put in a comment to thank you :) |
| 15:57.56 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 15:57.58 | jaytee | one thing I'd like to see that's not actually related to the process but I think would be really helpful to people considering Asterisk or already using it but considering upgrading is a document that all the features available in all releases and next to each feature the branch of Asterisk that the feature was first included. |
| 15:58.33 | seanbright | you should be able to compile that from CHANGES and UPGRADE-* relatively easily |
| 15:58.37 | jaytee | umm, should have said "with all the features" not that all the features |
| 15:58.44 | seanbright | it wouldn't be exhaustive by any means, but it would be a start. |
| 15:59.04 | jaytee | seanbright, yeah, I'll get right on that cuz all I have to do is housecleaning :-) |
| 15:59.11 | kaldemar | are all 1.6 branches going to be supported equally? |
| 15:59.28 | seanbright | kaldemar: until otherwise announced, yes. |
| 15:59.57 | seanbright | actually, that's a lie |
| 16:00.04 | seanbright | 3 branches |
| 16:00.10 | drmessano | Its noted in the post |
| 16:00.11 | seanbright | 1.6.0, 1.6.1, and 1.6.2 |
| 16:00.18 | seanbright | 1.6.1, 1.6.2, 1.6.3 |
| 16:00.19 | drmessano | and my paste |
| 16:00.21 | seanbright | etc, etc,. |
| 16:01.10 | drmessano | seanbright I do have one question |
| 16:01.30 | seanbright | drmessano: i most likely do not have the answer, but shoot. |
| 16:01.31 | kaldemar | the post says at least 3 branches, not 3 explicitly. |
| 16:02.14 | drmessano | Right now, 1.2 = Security fixes (EOL), 1.4 = Bugfixes, any 1.6 is fully supported |
| 16:02.18 | kaldemar | bugfixing effort may grow quite a bit if a lot of refactoring is made and there are many branches. |
| 16:02.23 | *** join/#asterisk af_ (n=getsmart@88-149-240-107.dynamic.ngi.it) |
| 16:02.31 | seanbright | kaldemar: yes, we're already experiencing that. |
| 16:03.19 | seanbright | drmessano: what is the difference between "bug fixes" and "fully supported" |
| 16:03.23 | drmessano | Moving to 1.6.3.x, does that put 1.6.0.x at EOL/Security fixes ? |
| 16:03.25 | *** join/#asterisk IPPBX-ARG (n=pirruar@200-112-152-64.bbt.net.ar) |
| 16:03.41 | *** part/#asterisk IPPBX-ARG (n=pirruar@200-112-152-64.bbt.net.ar) |
| 16:03.54 | drmessano | Well, seems some distinction is noted between 1.4's support and 1.6.x support |
| 16:03.58 | leifmadsen | drmessano: not necessarily |
| 16:04.12 | leifmadsen | drmessano: there is no difference between 1.4 and a 1.6 branch |
| 16:04.19 | drmessano | I see |
| 16:04.20 | leifmadsen | same methodology |
| 16:04.26 | *** join/#asterisk madfactor (i=madfacto@74-128-245-18.dhcp.insightbb.com) |
| 16:04.28 | *** join/#asterisk MT`AwAy (n=MagicalT@shigoto.ookoo.org) |
| 16:04.29 | seanbright | while getting the 1.6 releases polished is the focus, 1.4 is still getting bug fixes and is supported. |
| 16:04.34 | leifmadsen | bug fixes, security fixes -- no new features |
| 16:04.45 | leifmadsen | 1.4 is not going away anytime soon |
| 16:04.47 | seanbright | right. all release branches are feature frozen. |
| 16:04.54 | leifmadsen | no discussions to move it to EOL have been discussed |
| 16:05.02 | drmessano | ok |
| 16:05.09 | leifmadsen | and if such a time comes that 1.4 is EOL'd, you'll be given plenty of notice |
| 16:05.17 | seanbright | yeah |
| 16:05.17 | leifmadsen | i.e. several months to a year most likely |
| 16:05.18 | seanbright | at least a week |
| 16:05.20 | drmessano | I'm past 1.4.. so not worried about |
| 16:05.23 | leifmadsen | (that is my interpretation anyways :)) |
| 16:05.26 | MT`AwAy | hello, is there any change to the sip (sippeers, sipusers) realtime database between 1.4 and 1.6, and where can I find the changes if any? |
| 16:05.30 | leifmadsen | drmessano: but many would be :) |
| 16:05.43 | leifmadsen | MT`AwAy: I'd check UPGRADE-1.6.txt |
| 16:05.44 | rob0 | Let's discuss a discussion right now, he said, disgusted. |
| 16:05.45 | drmessano | Just curious how future was going to play out when say 1.6.0 is the 4th or 5th branch behind |
| 16:05.55 | leifmadsen | rob0: that's disgusting! |
| 16:05.57 | MT`AwAy | leifmadsen> already done, it doesn't speak much about realtime |
| 16:06.09 | leifmadsen | MT`AwAy: CHANGES ? |
| 16:06.26 | leifmadsen | drmessano: that has not been decided yet since we don't have a 4th branch yet |
| 16:06.41 | seanbright | god... that's going to be a total pain in the ass. |
| 16:06.42 | leifmadsen | I suppose it comes down to how much effort is required to maintain X number of branches |
| 16:06.52 | leifmadsen | seanbright: we need a better way of merging.... |
| 16:07.05 | leifmadsen | mergetrunk6 1.6.0 1.6.1 1.6.2 1.6.3 |
| 16:07.07 | seanbright | leifmadsen: i already use shell scripts :) |
| 16:07.21 | leifmadsen | seanbright: hawt :) share them with the world! :) |
| 16:07.22 | MT`AwAy | I'm currently trying to update to 1.6, but I became unable to call a user (behaviour looks like bug 13121) |
| 16:07.45 | seanbright | for x in 0 1 2; do pushd "1.6.${x}"; mergetrunk6 123456; svn resolve . ; svn commit -F ../merge.msg; popd; done |
| 16:07.48 | leifmadsen | ok, gotta go! thanks for all the feedback on the post everyone! next week will probably be something to do with Queue's per seanbright's suggestion |
| 16:08.02 | leifmadsen | seanbright: that's hawt |
| 16:08.05 | seanbright | heh |
| 16:08.09 | madfactor | Can ne1 direct me to any decent resources on meridianopt11-asterisk interop... besides what is available on voip-info.org? |
| 16:08.12 | seanbright | assuming there are are no conflicts, yes :) |
| 16:08.16 | seanbright | ok, go away |
| 16:08.19 | seanbright | right now |
| 16:08.39 | aliverius | i shall ask once more, do i still have to get the chan_misdn module from beronet? |
| 16:08.40 | leifmadsen | beer me! (several times!) |
| 16:09.35 | seanbright | aliverius: for which version of asterisk? |
| 16:10.22 | drmessano | IMHO, it shouldnt be that big of a deal.. Even though each release in 1.6 could/can/will be a leap from the previous, as you said, it's not like the 2 years between, and it should be easier to migrate up in a timely manner, much like (dare I say) service packs on M$ systems introduce some big changes, but really, it's not a giant leap |
| 16:10.38 | drmessano | heh |
| 16:10.56 | seanbright | not to mention - new features get out more quickly and have the kinks worked out |
| 16:11.05 | drmessano | Exactly |
| 16:11.11 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:11.23 | seanbright | i sometimes feel like the .0 releases are public QA releases... so.... |
| 16:11.26 | seanbright | thanks everybody! |
| 16:11.26 | seanbright | heh |
| 16:12.20 | drmessano | This release cycle is really much more down to earth than it's been before.. more inline with other applications where upgrading from 2.4 to 2.7 shouldnt kill any kittens |
| 16:12.48 | seanbright | i still would like to convince the other devs to do a 2.0 release |
| 16:12.57 | seanbright | which is a release where we really right all of the wrongs |
| 16:13.03 | seanbright | and the community will actually let us break BC |
| 16:13.33 | seanbright | aliverius: i shall ask once more, what version of asterisk? |
| 16:14.19 | drmessano | Really, this release cycle leaves a lot of room to stop babysitting those on 4 year old code... Its going to be as lot easier to tell someone "1.6.0.x is unsupported, please migrate to a newer release" than telling someone "1.2 is old, you need to be on 1.6" where a giant f*ck you was in order. |
| 16:14.56 | *** join/#asterisk hi365 (n=hi365@94.159.176.33) |
| 16:14.57 | seanbright | drmessano: i wish that were true |
| 16:15.19 | seanbright | there are those that think the dialplan they wrote in 1.0 should work in 1.6.2 |
| 16:15.20 | [TK]D-Fender | drmessano: Better answered as "1.2 is broken? Well it isn't going to get fixed, so pick your poison" |
| 16:15.28 | seanbright | and we are a bunch of idiots for not being able to make that work. |
| 16:15.29 | aliverius | seanbright: sorry for delaying i am messing in menus and docs and and :) |
| 16:15.37 | aliverius | v1.6.1.1 |
| 16:15.51 | seanbright | aliverius: well you seemed impatient, i was just responding in kind. |
| 16:15.58 | [TK]D-Fender | seanbright: Well with the deprecation cycle, so many users of 1.2 are using 1.0 dialplan and expecting it to work even in 1.4 |
| 16:16.00 | aliverius | :) |
| 16:16.03 | aliverius | ty |
| 16:16.30 | seanbright | aliverius: i see chan_misdn.c in the 1.6.1.1 release... so what are you asking? |
| 16:16.39 | seanbright | libmisdn? |
| 16:16.43 | aliverius | i see it there in menulist but then it seems it asks for some deps marked with (E) which should mean external i suppose? |
| 16:16.54 | seanbright | correct |
| 16:17.03 | aliverius | chan_misdn.so |
| 16:17.09 | drmessano | seanbright: I think the team should really consider that stance on supporting and endorsing the older code.. While I do think its unreasonable to tell someone they need to hurry up and move their 1.2 systems to 1.6.1.x tomorrow, it shouldnt/wont be to tell them the same for moving from the future unsupported 1.6.0.x to say 1.6.5.x |
| 16:17.19 | aliverius | it lists three deps |
| 16:17.42 | drmessano | Which will help the perception when it comes to deprecation and attrition |
| 16:17.43 | aliverius | isdnnet(E), misdn(E), suppserv(E) |
| 16:17.56 | aliverius | now i need to locate 2 of them... |
| 16:18.26 | seanbright | ISDN4Linux Library |
| 16:18.34 | seanbright | mISDN User Library |
| 16:18.55 | seanbright | mISDN Supplemental Services |
| 16:19.02 | seanbright | search for those on the interwebs |
| 16:19.09 | seanbright | what distro are you on? |
| 16:20.26 | drmessano | Going back to what I brought up before.. Telling someone to upgrade from Windows 2000 server to Windows 2003 server to fix an issue is nuts.. but it's perfectly reasonable to tell someone they're on SP0 and need to go ahead and upgrade to SP2.. With that mindset, I think folks will migrate faster to 1.6 knowing it will be easier to keep up |
| 16:20.42 | seanbright | i agree with you. |
| 16:20.46 | *** join/#asterisk haikiro (n=superhai@83-103-99-30.ip.fastwebnet.it) |
| 16:20.56 | aliverius | seanbright: archlinux |
| 16:21.04 | seanbright | aliverius: seriously? why god, why?? |
| 16:21.11 | aliverius | i installed misdn userscpace i thought those whwere enough |
| 16:21.32 | haikiro | hi all any italian user here ? |
| 16:21.33 | aliverius | for a fine control over my system ;) |
| 16:21.59 | aliverius | io non sono italiano haikiro, ma lo parlo |
| 16:22.13 | drmessano | But really.. thats NOT a technical issue... thats a perception issue that needs to be brought up over and over and over by the devs until its just "fact" amongst the users and admins out there.. |
| 16:22.19 | seanbright | aliverius: well you need those three packages. i don't know where you find them for your distro. |
| 16:22.40 | haikiro | aliverrius posso disturbarti in pvt? |
| 16:22.42 | seanbright | drmessano: well the third-party module issue always plays into that. and *is* a technical issue. |
| 16:22.43 | smps | seanbright, i use slackware , whats wrong with archlinux ? |
| 16:22.53 | aliverius | si haikiro |
| 16:23.02 | seanbright | not getting into a religious distro discussion |
| 16:23.08 | seanbright | i retract my statement about archlinux |
| 16:23.10 | seanbright | it's wonderful |
| 16:23.17 | seanbright | i use it myself |
| 16:23.37 | drmessano | seanbright: Of course.. and in my scenario, SP2 didn't play well with every app that worked on Server 2003 release.. so they plan a little, upgrade, no sweat.. not like 1.2 to 1.6 or even 1.2 to 1.4 |
| 16:23.42 | seanbright | long story short: deprecation is now a dirty word. |
| 16:24.00 | aliverius | hahaha, distro religion! |
| 16:24.06 | smps | hehe |
| 16:24.18 | aliverius | ok i can make my packages nice and easy IF i find their source |
| 16:25.19 | smps | aliverius, why dont you check misdn site you will find there explanations what all do you need for it to work ... |
| 16:25.29 | drmessano | But see, I think where there's been a BIG perception failure is that I have seen little to encourage the leap from 1.2/1.4 to 1.6 based on what the future will bring.. I think like Leif's post, there are good explanations how the new release cycle works, but nothing where its said "If you move to 1.6 now, it will be EASIER to keep up" |
| 16:26.26 | seanbright | but some would respond "i don't want to keep up! digium is a company with a product and should support it for all of us forever!" |
| 16:26.33 | seanbright | and my FAVORITE illogical argument: |
| 16:26.50 | drmessano | I see time and time again, the perception is "1.2 is fine.. if I go to 1.4 or 1.6.. I am going to have to redo everything.. why bother, i'll just have to do it again for the next release" |
| 16:26.57 | seanbright | "i am building a system that is supposed to run for 10+ years!" |
| 16:27.01 | seanbright | "... and i want to upgrade" |
| 16:27.13 | haikiro | aliverius leggi i miei mess in pvt? |
| 16:27.26 | seanbright | people are only encouraged to move off of older versions if they have a need |
| 16:27.31 | seanbright | run 1.2, i don't care |
| 16:27.43 | seanbright | but if you run into a bug, you either have to fix it, pay someone to fix it, or upgrade |
| 16:28.04 | seanbright | everyone is lazy |
| 16:28.05 | seanbright | heh |
| 16:28.08 | drmessano | I don't think thats the majority at all.. Everyone wants to keep up, but they dont want to WORK to do it.. and if they do need to rewrite their diaplans everytime they upgrade, they wont.. ever.. If its a sed now and then as they move up, or a minor change here and there, they will love it |
| 16:28.40 | seanbright | "vocal minority" |
| 16:28.41 | seanbright | :) |
| 16:28.49 | [TK]D-Fender | drmessano: Well in 1.2 you could use 1.0 dialplan that was being deprecated. By moving to 1.4 even though 1.6 is out there, at least you'll have access to 1.4 stuff (mostly) when you move to 1.6 |
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| 16:29.58 | smps | drmessano, i dont think its dialplan rewrite problem, it is problem when you have to test it all , and dont have that much resources or you have to implement that "test" system wihin clients and when something goes wrong you are then loosing clients .... |
| 16:30.12 | seanbright | well then you shouldn't be upgrading |
| 16:30.51 | smps | seanbright, its not problem , i can keep with that at home |
| 16:31.05 | drmessano | I think thats a bit illogical.. |
| 16:31.13 | smps | seanbright, but within job where i implement something like that i cant come and say wait i need something to test |
| 16:31.20 | [TK]D-Fender | smps: and you just said "test with clients". Thats like "cleaning loaded guns" |
| 16:31.31 | smps | hehehe |
| 16:31.39 | seanbright | you can't afford to test? |
| 16:31.42 | smps | [TK]D-Fender, thats what i mean |
| 16:31.48 | seanbright | note to self: don't contract with smps |
| 16:31.54 | smps | hahahaha |
| 16:31.55 | drmessano | I think the biggest lure of the 1.6 release cycle is being missed in how it's being promoted.. |
| 16:32.11 | smps | seanbright, yeah i cant pay bunch of isdn hardware and isdn line @home to test all of that stuff |
| 16:32.13 | smps | its insane |
| 16:32.49 | seanbright | asterisk is infinitely configurable, which makes it infinitely difficult to test during development, unfortunately. |
| 16:33.04 | seanbright | but that *is* being addressed. |
| 16:33.28 | smps | ok |
| 16:33.37 | seanbright | smps: run 1.0 |
| 16:33.40 | seanbright | that shit is rock solid |
| 16:33.47 | smps | seanbright, i do run 1.4 |
| 16:33.49 | seanbright | or run trunk with your hair on fire |
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| 16:34.30 | seanbright | i really need to buy a couple new boxes |
| 16:34.36 | seanbright | anyone want to sponsor asterisk dev? |
| 16:34.38 | seanbright | :) |
| 16:34.47 | smps | seanbright, where do you live |
| 16:35.04 | seanbright | baltimore, maryland, usa |
| 16:35.08 | seanbright | 'the old line state' |
| 16:35.12 | seanbright | don't ask me. |
| 16:35.14 | seanbright | shrugs |
| 16:35.30 | smps | huh a bit away from here |
| 16:35.33 | [TK]D-Fender | seanbright: Nothing so stable as a dead-stop :p |
| 16:36.01 | seanbright | [TK]D-Fender: feature creep is not a concern. |
| 16:37.54 | seanbright | some of the GSoC projects going on are going to make integrating with asterisk a treat |
| 16:38.14 | seanbright | data get, followed by data put |
| 16:38.41 | seanbright | someone has to come up with some kind of real-time instrumentation and tuning application |
| 16:39.06 | seanbright | and the event stuff russellb has been working on is pretty hot, too. |
| 16:40.30 | seanbright | oh, and we're rewriting the whole thing in python. |
| 16:48.37 | aliverius | "mISDN Supplemental Services" |
| 16:48.45 | aliverius | now wehat the hell are those? |
| 16:50.58 | seanbright | http://www.misdn.org/index.php/MISDN_with_Asterisk |
| 16:51.43 | seanbright | specifically, configure is looking for mISDNuser/suppserv.h |
| 16:51.50 | seanbright | so whichever package has that is the one you want |
| 16:56.45 | aliverius | it seems i will have to download from beronet even with asterisk 1.6? |
| 16:56.46 | aliverius | anyway |
| 16:56.50 | aliverius | thanks a lot! |
| 16:56.58 | aliverius | i will try later again, gtg for now |
| 16:57.02 | seanbright | enjoy. |
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| 17:06.10 | metfan2007 | Hi all! after implementing some macros in my dialplan I'm getting the "h" extensions in all my dst cdr field, the agents dials a number, I don't have any Set(CDR(dst)) in my dialplan, I need the number dialed |
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| 18:31.43 | TerrenceLam | +C |
| 18:32.03 | TerrenceLam | C |
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| 19:33.10 | rjune | I am trying to setup direct dial for a particular line on a Digium card. Is there anything specific telco needs to support for this to work? or should I simply attach a particular port to that extension? |
| 19:36.53 | [TK]D-Fender | rjune: meaning? |
| 19:37.28 | rjune | Meaning I want to dial a number and always have a particular extension ring |
| 19:38.00 | [TK]D-Fender | rjune: well if you can dial a specific # to call that analog line, the * can process the call any which way you want. |
| 19:38.11 | [TK]D-Fender | rjune: The telco is not incontrol of how you want to treat your calls |
| 19:38.58 | rjune | No but I keep seeing a reference to a DID from the trunk, I wasn't sure if the digium card provided information on the number being called vs the number calling in or not |
| 19:39.25 | rjune | aka, I'm not 100% sure where the DID comes from. |
| 19:39.31 | [TK]D-Fender | rjune: Well you haven't told us what card you are using, or what signalling on it |
| 19:40.41 | rjune | it's a digium wildcard tdm800p |
| 19:40.54 | rjune | FXS signalling |
| 19:41.11 | [TK]D-Fender | rjune: Analog does not have the concept of DIDs |
| 19:41.19 | [TK]D-Fender | rjune: Its just a dumb ringing line |
| 19:41.36 | rjune | ok. so I have to bind a given port on the card to the extension |
| 19:42.01 | [TK]D-Fender | rjune: You could tell your telco 3 different #'s or reasons you might want a call to land on it, but no device will have any clue which. |
| 19:42.18 | [TK]D-Fender | rjune: And I'd avoid terms like "bind" you seem to throw around |
| 19:42.33 | rjune | what term would you prefer? |
| 19:42.39 | [TK]D-Fender | rjune: Makes things sound like there is some magical jack for things. |
| 19:42.45 | [TK]D-Fender | rjune: Its all just dialplan. |
| 19:43.13 | [TK]D-Fender | rjune: Nothing requires you to have a call coming in on any one interface to be treated like any other |
| 19:44.28 | rjune | I have the dialplan configured so that generically if a call comes in, it rings extension X |
| 19:44.51 | rjune | I have a phone line I want all calls on that line to ring extension Y instead of X |
| 19:45.33 | [TK]D-Fender | rjune: You control where every port on your card goes in the dialplan. |
| 19:45.56 | rjune | it's a different dialplan to do that, and I have to set the dialplan up such that it uses a particular port on the card, yes? |
| 19:46.27 | rjune | There's nothing magical about it, bind or tie just seemed to accurately describe what I need to do. |
| 19:46.28 | [TK]D-Fender | rjune: Dialplan using a port on the card? thats OUTBOUND <- |
| 19:46.53 | rjune | I can't setup an inbound dialplan for a particular port? |
| 19:47.19 | [TK]D-Fender | rjune: ... each port can be pointed to whatever context you want. Whatever is waiting in there is up to you |
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| 19:59.38 | sfire | I am trying to configure asterisk without using a GUI. I get the SIP clients registered and I get the trunk registered but it spits out errors on incoming calls that it cannot find my extension and errors on outgoing calls saying that it cannot find the extension (it thinks the phone number is an extension) |
| 19:59.56 | sfire | anyone have a link to a good guide or something that could help me to fix this? |
| 20:01.07 | [TK]D-Fender | sfire: If you've already read the book and looked around then the docs aren't going to fix your misconceptions. |
| 20:01.22 | [TK]D-Fender | sfire: Pastebin your failed call and your dialplan and we'll show you where you went wrong. |
| 20:01.30 | [TK]D-Fender | sfire: make sure SIP DEBUG is enabled |
| 20:01.32 | [TK]D-Fender | !pb |
| 20:01.36 | [TK]D-Fender | ~pb |
| 20:01.37 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 20:02.33 | [TK]D-Fender | sfire: And an extension IS a phone number. or better worded a number dialed on a phone. What it does is unimportant |
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| 20:14.30 | rjune | [TK]D-Fender, Thanks, you clarified where I needed to look and it's working. |
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| 20:23.45 | iksik | hmm |
| 20:24.53 | iksik | h323 and SIP can run together? |
| 20:25.41 | RyanRR | ... |
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| 20:34.57 | [TK]D-Fender | iksik: They have nothing to do with each other |
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| 20:45.53 | buttons840 | I get a segmentation fault when starting asterisk http://pastebin.com/d62b5e0c2 i don't even know where to begin troubleshooting... |
| 20:48.16 | iksik | [TK]D-Fender, hmm, damn it :/ |
| 20:48.20 | iksik | damn debian ;/ |
| 20:51.11 | Maliuta | iksik: how so? |
| 20:51.43 | Maliuta | iksik: and I think tzafrir_laptop might have something to say about that statement, since he is partially responsible for the packaging |
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| 20:52.34 | iksik | great |
| 20:53.14 | tzafrir_laptop | hmm... what's the problem exactly? |
| 20:53.46 | tzafrir_laptop | (asterisk-h323 is in a separate package because it pulls the extra big openh323 dependency) |
| 20:53.53 | buttons840 | is dahdi-linux a distro or a driver? |
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| 21:03.44 | iksik | tzafrir_laptop, ok, how can I make it works "out of the box" together ? |
| 21:04.08 | tzafrir_laptop | just install asterisk-h323 in addition to asterisk |
| 21:04.19 | tzafrir_laptop | (which is a dependency of asterisk-h323 anyway) |
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| 21:07.21 | [TK]D-Fender | buttons840: the kernel driver for DAHDI |
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| 21:19.05 | buttons840 | so, this segementation fault in asterisk, could it be cause by outdated zaptel drivers? |
| 21:25.23 | wdoekes | buttons840: attaching a backtrace helps |
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| 21:26.17 | wdoekes | (gdb /my/asterisk <enter> run my-argument-list <enter> <segfault> bt <enter>) |
| 21:26.35 | iksik | tzafrir_laptop - asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext |
| 21:26.40 | iksik | asterisk from package :| |
| 21:26.55 | tzafrir_laptop | what package is it? what version? |
| 21:27.42 | tzafrir_laptop | this is a symbol that should come from asterisk itself |
| 21:27.42 | iksik | how can I check it ? |
| 21:27.46 | iksik | yeap |
| 21:28.04 | tzafrir_laptop | what package is it? what version? |
| 21:28.04 | iksik | (this is not mine :D) |
| 21:28.04 | iksik | package: asterisk ;> |
| 21:28.04 | tzafrir_laptop | dpkg -l asterisk\* | grep ^i |
| 21:28.18 | iksik | ii asterisk 1:1.4.21.2~dfsg-3 |
| 21:31.53 | buttons840 | wdoekes, what is gdb? |
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| 21:32.11 | wdoekes | gnu debugger |
| 21:34.47 | buttons840 | let me install and try it |
| 21:35.08 | wdoekes | iksik: do you have a features.conf ? |
| 21:35.32 | buttons840 | wdoekes, what is "gnu debugger" |
| 21:35.40 | buttons840 | nm |
| 21:35.59 | iksik | wdoekes no |
| 21:37.10 | wdoekes | ast_pickup_ext is defined in res_features.so, make sure it's loaded |
| 21:38.01 | iksik | hum, ok |
| 21:39.23 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 21:40.01 | wdoekes | I'm off, good luck all |
| 21:41.35 | iksik | ok, pbx.c:2981 ast_register_application: Already have an application 'Directory' |
| 21:41.41 | iksik | :| |
| 21:42.23 | iksik | how to fix it ? |
| 21:43.06 | barbacha | [TK]D-Fender: ok, I finished reading "* the future of VoIP" and I'm still not quite sure on how you would bridge a T2. Chapter 15 seems to say it's possible / working but I found no details on *how* |
| 21:44.13 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:44.27 | barbacha | [TK]D-Fender: would this be simply some rules in the dialplan (extensions) ? |
| 21:45.32 | buttons840 | speaking of asterisk the future of voip, would you suggest i read the whole book through once first, or should i try to fallow along the first read? |
| 21:45.33 | [TK]D-Fender | barbacha: T2.... sure you're not meaning T1 ? |
| 21:45.43 | barbacha | [TK]D-Fender: I am |
| 21:46.03 | [TK]D-Fender | barbacha: And I explained this already, there is no miracle "bridge". Call comes in A. your dialplan will simply DIAL out B |
| 21:46.21 | MT`AwAy | anyone knows where output of ast_debug(1, "text", ...) is sent (asterisk 1.6.1.0) ? I tried to logger set level debug on but got nothing |
| 21:46.33 | [TK]D-Fender | barbacha: "am" what? Meaning T1? Or that you actually have a T2. I've never seen anyone with a circuit like that here |
| 21:47.08 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:47.11 | RyanRR | t2??? |
| 21:47.14 | RyanRR | I think you're wrong |
| 21:47.19 | RyanRR | maybe you're thinking of two t1's bonded |
| 21:47.26 | iksik | damn it, i've added noload => app_directory, and now i've got 4 other errors - with VoiceMail, VoiceMailMain, MailboxExists and VMAuthenticate |
| 21:47.27 | iksik | ;/ |
| 21:48.10 | barbacha | why should I be wrong ? T2 exists right ? What do you plug on a Digium TE220B ? |
| 21:48.19 | RyanRR | because nobody uses t2 |
| 21:48.22 | *** join/#asterisk telecos (n=sergio@91.166.219.87.dynamic.jazztel.es) |
| 21:48.45 | barbacha | I have to admin I'm not quite familiar with all those terms yet |
| 21:48.50 | [TK]D-Fender | barbacha: You pluge T1's into Digium cards |
| 21:48.57 | barbacha | ok my bad then |
| 21:49.16 | [TK]D-Fender | barbacha: So I've answered this about 3 times now, and the answer isn't changing. |
| 21:49.33 | barbacha | [TK]D-Fender: so this is all about dialplan then |
| 21:49.54 | barbacha | [TK]D-Fender: setting up extensions that Diall($OLDNUM) |
| 21:51.57 | [TK]D-Fender | barbacha: go install * and a softphone and actually try using it. |
| 21:52.23 | barbacha | :) |
| 21:52.27 | barbacha | I did |
| 21:52.49 | [TK]D-Fender | barbacha: Dial() <- thats almost all you have to do to spit a call in from A out to B |
| 21:52.52 | barbacha | and actualy maintain the *s at my company (5 of them with IAX and the whole shit) |
| 21:53.13 | barbacha | but I'm new to * I admin |
| 21:53.16 | [TK]D-Fender | barbacha: Then you should already know that every call is just a stupid call. In or out makes no difference |
| 21:59.42 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 22:00.55 | nextime | hello all. On a little atom 1.6 gigs dual core cpu with only 512 megs of ram, with * 1.6 latest, how many users can i expect as limit for a meetme conference givin that i have enough bandwidth? |
| 22:02.39 | *** join/#asterisk DarkRift (n=dark@65.92.249.175) |
| 22:07.28 | buttons840 | wdoekes, I can't figure out how to use gdb. I've installed it, but how to I use it to run asterisk? |
| 22:09.34 | *** join/#asterisk aliverius (n=aliveriu@chal530-a049.home.otenet.gr) |
| 22:12.13 | iksik | ok |
| 22:12.15 | iksik | ERROR[3720]: chan_h323.c:3186 load_module: Gatekeeper registration failed. |
| 22:12.31 | iksik | how should looks like working entry of GK user ? :| |
| 22:13.05 | *** join/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 22:13.48 | buttons840 | http://pastebin.com/d2a925213 - ok wdoekes this is the back trace. |
| 22:16.01 | iksik | anyone can help me with GK settings? :( |
| 22:16.20 | Maliuta | google? |
| 22:16.36 | Maliuta | do we have a trigger for that? |
| 22:16.39 | Maliuta | ~google |
| 22:16.40 | infobot | extra, extra, read all about it, google is http://lmgtfy.com/?q=google |
| 22:19.21 | RyanRR | man you guys are fucking dicks |
| 22:19.28 | RyanRR | do you get off on being here or something lol |
| 22:20.12 | MT`AwAy | I'm trying to get video working with asterisk 1.6 - when I was on 1.4 running the "echo test" was also displaying video as echo, is it still the case on 1.6 ? |
| 22:20.51 | MT`AwAy | seems that it should still be the case |
| 22:21.02 | iksik | Maliuta, what exactly on google? ;> |
| 22:23.21 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 22:24.32 | wwalker | for best performance nad audio quality, if I use g711u, I want all my recordings in .ul format, right? |
| 22:24.35 | wwalker | or wrong? |
| 22:24.49 | wwalker | s/nad/and/ |
| 22:26.30 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) [NETSPLIT VICTIM] |
| 22:26.30 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) [NETSPLIT VICTIM] |
| 22:26.30 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 22:26.30 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 22:27.44 | MT`AwAy | http://pastebin.com/m4fa43470 <- is it normal that "Video Support" is displayed as "no" when format includes h263p and h264 ? |
| 22:30.37 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 22:30.57 | rue_mohr | hmm so phone audio works ok ulaw at 64K |
| 22:31.09 | rue_mohr | a serial port can do 115K |
| 22:31.15 | rue_mohr | hmmm |
| 22:32.41 | rue_mohr | ata can equiv do 2.1Gbigs |
| 22:32.45 | rue_mohr | hmmm |
| 22:33.12 | rue_mohr | Gbits :) |
| 22:33.44 | [TK]D-Fender | MT`AwAy: did you put "videosupport=yes" in [general]? |
| 22:33.55 | rue_mohr | so its totally possable to make a single port card that runs on a serial port, or run more than a whole channelbank off an ata port |
| 22:34.28 | rue_mohr | so I could make a ulaw channelbank that runs on an ide port |
| 22:35.06 | MT`AwAy | [TK]D-Fender> I did |
| 22:35.19 | MT`AwAy | even restarted asterisk as I wasn't sure "reload" had been effective |
| 22:35.45 | [TK]D-Fender | MT`AwAy: pastebin the complete call with SIP debug, and your configs |
| 22:36.27 | buttons840 | anyone familiar with libvpb0 ? having this installed causes a seg fault when starting asterisk, so i removed it, but what does it do? |
| 22:36.56 | rue_mohr | I shoudl be darring and totally upgrade my home install |
| 22:39.40 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 22:42.25 | [TK]D-Fender | rue_mohr: go WILD... |
| 22:47.29 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:50.09 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:51.38 | C4colo | can asterisk play a file from an http stream? |
| 22:51.54 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:52.23 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 22:52.29 | C4colo | Playback(http://server/script?tts=something+needs+to+be+said+using+tts) ... or something like that |
| 22:52.46 | C4colo | or is there an RPC playback function that I don't know about? |
| 22:53.48 | C4colo | or can I do this using AGI or AMI? |
| 23:11.47 | *** join/#asterisk bbryant1 (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 23:14.18 | wwalker | I've got ulaw sound files and I'm talking g711u to my provider, but asterisk says it can't find my sound files. show file formats doesn't show ulaw, what do I need to load for it to support ulaw (trying to decrease the transcoding)? |
| 23:15.06 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:17.12 | [TK]D-Fender | wwalker: Show us the call with SIP debug enabled, and an "ls -la" of your sounds folders |
| 23:19.12 | wwalker | [TK]D-Fender: I just needed to know whether it was the codec for ulaw or which format file (since there is no format for ulaw |
| 23:19.26 | wwalker | turns out format_psm.so includes ulaw |
| 23:44.13 | *** join/#asterisk medavian (n=darrink@203.80.168.222) |
| 23:45.16 | *** part/#asterisk medavian (n=darrink@203.80.168.222) |