| 00:13.38 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 00:32.43 | edong23 | hackish: some devices have a dialplan inside them |
| 00:43.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:49.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:49.38 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:49.38 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:54.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:58.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:00.21 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 01:05.03 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:16.51 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:23.57 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:35.54 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 01:46.14 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:46.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:56.30 | BludSuckingFiend | aah, fun with the PITCH_SHIFT function |
| 01:56.53 | BludSuckingFiend | puts it in the CEO's conference bridge |
| 02:09.06 | *** join/#asterisk PrestonL (~preston@206.81.170.217) |
| 02:14.20 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:18.01 | hackish | I just found that primus has a tool to make config files for me. they're for 1.4 but I assume these will still work with 1.8. Syntax looks to be about the same. |
| 02:19.21 | *** join/#asterisk tengulre (~tengulre@182.148.109.115) |
| 02:21.25 | edong23 | hackish: just write them |
| 02:21.41 | edong23 | you will learn more, and youll end up with cleaner configs |
| 02:21.44 | edong23 | well, maybe... |
| 02:21.53 | tengulre | hi,all |
| 02:22.27 | hackish | Honestly I think using their generator is a good idea for me because otherwise I have no way of knowing what options they'd support. a) get it working, b) play with it |
| 02:23.54 | hackish | unless these servers are more permissive than I think |
| 02:27.06 | hackish | 4569 is IAX - that's only UDP on both sides of the connection? |
| 02:32.12 | jero | yes |
| 02:32.40 | *** join/#asterisk poggio (~poggio@c-98-212-1-254.hsd1.il.comcast.net) |
| 02:36.05 | hackish | the defaults they give are this: iaxthreadcount = 2000 and iaxmaxthreadcount = 4000 |
| 02:36.18 | hackish | Seriously? What do you need 2000 threads for on a 2 line system? |
| 02:45.10 | BludSuckingFiend | because that gives you 1,000 threads per line |
| 02:45.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ibjngbevyykysede) |
| 02:45.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:49.25 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:49.25 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:49.27 | BludSuckingFiend | I've always preferred IAX to SIP for trunking... |
| 02:49.38 | BludSuckingFiend | Yay for efficient binary protocols. |
| 02:50.11 | hackish | sure. still getting the number you've dialled is not in service. |
| 02:54.00 | *** join/#asterisk jsarrel (~jsarrel@66-169-1-114.dhcp.gnvl.sc.charter.com) |
| 03:01.13 | hackish | I don't think this thing is registering. would that be iax2 show registry or iax2 show users? |
| 03:04.15 | BludSuckingFiend | show register I'd assume if it's registering with you |
| 03:04.22 | BludSuckingFiend | if you're registering with it... I'd think users |
| 03:04.38 | dr0ck | show registry is for your outbound registrations (their status) |
| 03:05.17 | dr0ck | use show peers..you can also 'database show' |
| 03:06.26 | *** join/#asterisk jsarrel (~jsarrel@66-169-1-114.dhcp.gnvl.sc.charter.com) |
| 03:06.45 | hackish | 0 iax2 peers [0 online, 0 offline, 0 unmonitored] |
| 03:07.25 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 03:07.28 | BludSuckingFiend | I assume they are in yout iax.conf and you've done an iax reload? |
| 03:07.44 | hackish | yes. I even tried a complete restart for good measure. |
| 03:07.57 | hackish | The iax.conf was based off the example they provided. |
| 03:08.25 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 03:08.57 | BludSuckingFiend | try turning on the what's the type? |
| 03:08.58 | hackish | the only thing I found odd was: register => 6133695073:mypassword@iax05.unlimitel.ca |
| 03:09.00 | BludSuckingFiend | peer, user or friend? |
| 03:09.35 | hackish | lemme pastebin it |
| 03:10.36 | hackish | http://pastebin.com/yi2V6w9Z |
| 03:15.34 | BludSuckingFiend | try setting it up as a friend |
| 03:16.01 | hackish | sure, I'll try that... that file is nearly identical to the sample they provided. |
| 03:18.28 | hackish | that seems to have done something...6133695073 0.0.0.0 (S) 0.0.0.0 4569 Unmonitored 1 iax2 peers [0 online, 0 offline, 1 unmonitored] |
| 03:19.49 | hackish | the DID rings now... doesn't connect but at least it doesn't say line not in service so I feel like it's progressing. |
| 03:19.52 | BludSuckingFiend | turn on the iax debug and see if you can see registration attempts? |
| 03:20.11 | BludSuckingFiend | add a host= line to it as well |
| 03:20.15 | BludSuckingFiend | the friend entry |
| 03:21.29 | hackish | No such command 'iax2 debug' (type 'core show help iax2 debug' for other possible commands) |
| 03:21.43 | BludSuckingFiend | iax2 set debug? |
| 03:21.58 | BludSuckingFiend | iax2 set debug on |
| 03:22.16 | hackish | aha. dox are wrong :( |
| 03:23.37 | hackish | strangely last time I had problems with sip I ran set debug on and it suddenly started working. Same deal here. |
| 03:24.02 | hackish | Soon as I ran that I called back. no other changes to the asterisk and this time the local phone rings... |
| 03:24.33 | BludSuckingFiend | maybe it took time to register |
| 03:24.49 | BludSuckingFiend | awhile back I had a trunk with a provider |
| 03:24.54 | BludSuckingFiend | I kept restarting asterisk |
| 03:24.59 | BludSuckingFiend | so it kept re-registering |
| 03:25.06 | BludSuckingFiend | their service blocked me for re-registering too much |
| 03:25.13 | BludSuckingFiend | for about 10 minutes |
| 03:25.34 | hackish | that could have been it too. |
| 03:25.57 | hackish | Do you think that changing type from user to friend was the key or just being more patient than I am? |
| 03:26.51 | BludSuckingFiend | well, user usually means incoming only if I recall correctly |
| 03:27.05 | BludSuckingFiend | if you want to send calls out you need it as a peer or friend |
| 03:27.28 | hackish | ok... either way I was trying to get incoming to work. |
| 03:34.03 | hackish | I think I must be missing a module or something. I tried setting autoload=yes and it registered but I set autoload=no and it seems to be just sending out a bunch of REGREQ packets with nothing coming back. |
| 03:34.43 | hackish | load => chan_iax2.so |
| 03:34.55 | hackish | don't know what else it would require |
| 03:36.29 | BludSuckingFiend | are you behind a NAT router? |
| 03:36.38 | *** join/#asterisk voxter_ (~voxter@d23-16-70-150.bchsia.telus.net) |
| 03:36.56 | BludSuckingFiend | because by default iax is UDP... you'll have to make sure you have the UDP port forwarded |
| 03:40.21 | hackish | I am behind a firewall but I set it up to forward UDP to that host. |
| 03:41.13 | BludSuckingFiend | odd |
| 03:41.33 | hackish | I'm assuming it's a module thing. |
| 03:41.55 | hackish | I can try setting autoload to yes again and restarting. If it works again it's confirmed. |
| 03:41.55 | BludSuckingFiend | I wouldn't think so... sounds like the other end simply isn't responding? |
| 03:42.16 | hackish | rdr xl0 0.0.0.0/0 port 4569 -> 192.168.0.38 port 4569 udp |
| 03:42.41 | hackish | been years since I set one of those up. it's possible that I need a keep state or something on it but I wouldn't think so. |
| 03:43.09 | hackish | given that all incoming UDP to port 4569 should be sent to the asterisk server |
| 03:43.24 | BludSuckingFiend | yeah |
| 03:43.32 | BludSuckingFiend | and if you got an incoming call before... |
| 03:43.35 | BludSuckingFiend | that's probably working |
| 03:43.44 | BludSuckingFiend | weird that you wouldn't be getting a response |
| 03:44.10 | hackish | restarting it with autoload on we'll see what happens. |
| 03:44.11 | BludSuckingFiend | maybe you need to set externip ? |
| 03:44.19 | BludSuckingFiend | due to the NAT |
| 03:45.31 | hackish | definitely some module thing. |
| 03:45.45 | hackish | restarted it and it seems to have registerred |
| 03:46.05 | hackish | phone call worked. |
| 03:49.15 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 03:49.54 | ectospasm | IAX2 maintains a connection once it's registered, you shouldn't have to configure any NAT-related settings in iax.conf and friends. |
| 03:50.30 | hackish | the only thing I see is if I set autoload=yes it works autoload=no it does not |
| 03:50.38 | BludSuckingFiend | makes sense with registration |
| 03:50.49 | hackish | did it twice with consistent results. |
| 03:51.26 | BludSuckingFiend | probably should see what the differences in the modules that are loading are |
| 03:51.55 | hackish | trying to find how you list the modules. 1/2 the commands you google up don't work. |
| 03:52.19 | hackish | similar to the iax2 set debug on thing. That's not what 90% of the websites say. |
| 03:52.26 | hackish | I assume it's been changed over the versions? |
| 03:53.22 | BludSuckingFiend | module show like chan_ |
| 03:53.29 | BludSuckingFiend | I can't find a way to show * |
| 03:53.29 | hackish | ok module show has a use number on the far right. |
| 03:53.43 | hackish | module show |
| 03:53.51 | hackish | helps when I don't type modules show |
| 03:54.02 | BludSuckingFiend | also the /var/log/asterisk/full |
| 03:54.07 | BludSuckingFiend | if you have it enabled |
| 03:54.38 | BludSuckingFiend | it will show each module it loads |
| 03:54.47 | hackish | looks suspiciously at res_agi.so Asterisk Gateway Interface (AGI) 1 |
| 03:55.14 | BludSuckingFiend | that's just for applications |
| 03:55.25 | BludSuckingFiend | like accessing asterisk from a perl script |
| 03:55.28 | BludSuckingFiend | not what it sounds like |
| 03:56.01 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.182) |
| 03:56.06 | BludSuckingFiend | acl |
| 03:56.27 | hackish | E486: Pattern not found: acl |
| 03:56.47 | hackish | res_timing_pthread.so pthread Timing Interface 1 |
| 03:56.55 | BludSuckingFiend | odd, it's in the list |
| 03:56.57 | hackish | that's another one I wasn't loading. |
| 03:57.05 | BludSuckingFiend | could be... |
| 03:58.17 | hackish | Do you think it would be better to just leave it loading 500mb of modules, get the machine running the way I want, then look at the module list and see what's being used and eliminate everything else? |
| 03:58.43 | hackish | it would be cool if you could just have it load things on demand. |
| 03:59.27 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-aliblhbzbewnmlal) |
| 03:59.38 | WIMPy | Might just be happening, even if manually. |
| 04:00.09 | BludSuckingFiend | I'd exclude what you know you don't need |
| 04:00.17 | BludSuckingFiend | I think there's an option to skip modules |
| 04:00.37 | BludSuckingFiend | noload => chan_h323.so |
| 04:00.43 | BludSuckingFiend | like that in modules.conf |
| 04:01.08 | hackish | honestly I don't know what 1/2 this stuff does. The descriptions in the book aren't even that helpful |
| 04:01.40 | BludSuckingFiend | yeah, really depends on what you use. |
| 04:02.07 | hackish | at 700mb it's a monster |
| 04:02.11 | hackish | asterisk 79088 0.0 1.2 695224 48364 ?? Is 10:44PM 0:47.33 /usr/local/sbin/asterisk -n -U asterisk |
| 04:02.46 | BludSuckingFiend | wow |
| 04:03.04 | BludSuckingFiend | mine only takes ~101M |
| 04:03.19 | hackish | that's with every single module loaded. |
| 04:03.41 | BludSuckingFiend | mine auto-loads too |
| 04:03.51 | BludSuckingFiend | so it has nearly everything loaded |
| 04:04.21 | BludSuckingFiend | It also has 4 full PRIs and 4 analog ports |
| 04:04.30 | BludSuckingFiend | and 140 SIP phones registered |
| 04:05.08 | BludSuckingFiend | and a monster of a dialplan |
| 04:06.50 | hackish | never know. It's built as a port so for all I know they built it with -g and no optimizations. |
| 04:07.41 | artyx | back it up, and install * on top? |
| 04:08.44 | hackish | you running a 64 bit OS bludsuckingfiend? |
| 04:09.02 | artyx | <PROTECTED> |
| 04:09.38 | artyx | Thats my 32.. on my 64 bit i have 15921 asterisk 20 0 2522m 30m 9.8m S 0.3 0.8 10:31.68 asterisk |
| 04:09.44 | artyx | but its a significantly smaller dialplan |
| 04:10.28 | hackish | that's 30mb? |
| 04:10.46 | hackish | or 2.5gb of ram? |
| 04:11.46 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 04:12.05 | hackish | disabled some stuff and in top it brought it down... 79307 asterisk 34 52 0 574M 42892K uwait 0 0:37 0.00% asterisk |
| 04:12.20 | hackish | exe also isn't stripped... I don't remember if that makes a diff or not. |
| 04:13.17 | hackish | none of the shared objects are stripped either... |
| 04:14.55 | hackish | no difference. |
| 04:19.49 | BludSuckingFiend | root 7605 13.4 76.6 3145724 1588988 pts/12 Dl 22:10 9:19 /usr/sbin/asterisk -f -vvvg -c |
| 04:19.57 | BludSuckingFiend | and yeah, I know, running as root bad... |
| 04:20.11 | BludSuckingFiend | But that was asterisk yesterday when it got in a loop |
| 04:20.20 | BludSuckingFiend | ~3GB |
| 04:20.44 | *** join/#asterisk nightrid3r (~kvirc@62.205.67.61) |
| 04:21.12 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 04:21.52 | hackish | I think my fault might be in who setup this port. seems to be compiling at least some things with -g. Again I'm not sure if the debug symbols are loaded into ram. |
| 04:25.04 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 04:28.52 | hackish | dahdi, I don't think I need that... don't have any digium cards. |
| 04:29.18 | hackish | with an ata box the only thing that talks does so over ethernet. |
| 04:32.20 | BludSuckingFiend | using meetme? |
| 04:32.45 | ectospasm | ConfBridge doesn't depend on MeetMe, IIRC |
| 04:32.56 | ectospasm | s/MeetMe/DAHDI/ |
| 04:33.29 | *** join/#asterisk brookshire (mbrooks@hijacked.us) |
| 04:33.36 | BludSuckingFiend | no, but MeetMe does... or did |
| 04:33.57 | BludSuckingFiend | I'll probably be looking at moving from MeetMe to Confbridge soon though |
| 04:37.16 | hackish | bad when I have to google meetme to know what it is... |
| 04:38.02 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 04:38.56 | hackish | extensions.conf http://pastebin.com/0L9hfhc7 |
| 04:39.30 | hackish | This thing seems pretty repetitive. Is there a way to automate a lot of these steps? Like for example they have a zillion entries for Set(CALLERID(all)=callerid <6133695073>) |
| 04:40.18 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 04:41.15 | hackish | I'm assuming for example exten => _NXXNXXXXXX,n,Goto(out1-${DIALSTATUS},1) could be rewritten to use the same exten as the other 10 rules? |
| 04:43.23 | BludSuckingFiend | yeah, though those separate out call types |
| 04:44.26 | hackish | what about the callerid lines? |
| 04:44.59 | BludSuckingFiend | lol, I saved 5M by stopping a bunch of stuff I don't need from loading |
| 04:45.39 | BludSuckingFiend | you could... but they are part of the other blocks |
| 04:45.46 | BludSuckingFiend | I would't bother consolidating just that line |
| 04:46.15 | BludSuckingFiend | that's a pretty small dialplan already |
| 04:46.39 | hackish | the reason I ask is only that this DID is only a temp until my number gets ported over. |
| 04:46.48 | hackish | search and replace works I suppose |
| 04:46.53 | BludSuckingFiend | my extensions.conf and includes are 8,000 lines |
| 04:47.13 | BludSuckingFiend | oh in that case, you could just use a global var |
| 04:48.24 | hackish | exten => _out1-.,1,Hangup(16) necessary? |
| 04:48.29 | hackish | it's a catchall no? |
| 04:50.36 | BludSuckingFiend | sort of yeah |
| 04:50.52 | BludSuckingFiend | only called if the dialstatus is something not recognized |
| 04:51.45 | hackish | that's what I thought. |
| 04:52.24 | hackish | No, moving the auth info stuff over. Is there an easier way to redo this: exten => _NXXNXXXXXX,n,Dial(IAX2/6133695073:passwd@iax05.unlimitel.ca/${EXTEN},60) |
| 04:54.21 | BludSuckingFiend | you should be able to use your peer/friend |
| 04:54.58 | BludSuckingFiend | IAX2/friendname/${EXTEN} I believe... for the dial string |
| 04:54.59 | hackish | ah yes, that makes sense. user was for incoming so they have to specifically authorize every single outgoing extension |
| 04:55.58 | *** join/#asterisk tengulre (~tengulre@182.148.109.115) |
| 04:56.32 | *** join/#asterisk stev3n (~Stiv@sever-kirov.ru) |
| 04:56.39 | stev3n | hi, about linux kernel options for asterisk, is this still actual? |
| 04:56.39 | stev3n | +# CONFIG_HZ_100 is not set |
| 04:56.39 | stev3n | +CONFIG_HZ_1000=y |
| 04:56.39 | stev3n | +CONFIG_HZ=1000 |
| 04:59.09 | drmessano | huh? |
| 05:00.44 | stev3n | I asked a funny question? |
| 05:04.25 | BludSuckingFiend | That question Hertz my eyes |
| 05:04.37 | hackish | lol |
| 05:05.32 | drmessano | Are you asking if CONFIG_HZ_1000=Y is still suggested/required for Asterisk? |
| 05:05.48 | drmessano | If so, then no |
| 05:07.13 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 05:19.33 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 05:28.44 | *** join/#asterisk resist0r (~resist0r@174.57.0.74) |
| 05:40.01 | *** join/#asterisk srp_ (~sandeep@115.119.115.26) |
| 05:55.01 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.107) |
| 05:55.06 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 05:56.26 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:56.31 | ruben231 | hi guys...any suggestion on best deployment setup if possible when my asterisk server will be doing calls different localtion around teh globe, namely, Australia, UK and the person will bedoing call comes from teh philippines, currently my setup is US server only- issue with disconnection. |
| 06:04.41 | edong23 | NOT AT ALL an asterisk question, but since this is a room full of voip guys, im curious: any of you messed with YATE? |
| 06:27.21 | hackish | If a call arrives, and you call Dial to send it to a certain phone, at what point does Dial return and processing move on to the next item in the dialplan? Ie if they answer it will the system sit in Dial until they hang up? |
| 06:28.19 | [TK]D-Fender | hackish: If they answer, yes |
| 06:29.46 | hackish | ok, I'm looking at calling Monitor on an extension. I assume I need to do this before calling dial then maybe delete the file if it returns BUSY or NOANSWER? |
| 06:29.53 | *** join/#asterisk tengulre (~tengulre@182.148.109.115) |
| 06:30.00 | tengulre | hi,all |
| 06:30.36 | [TK]D-Fender | hackish: You know there is a magical option to actually only begin recording when the call is BRIDGED.... |
| 06:31.00 | hackish | I did not |
| 06:31.41 | hackish | is that in monitor or mixmonitor? |
| 06:31.43 | [TK]D-Fender | hackish: You reaslly should read all of these apps instructions... |
| 06:31.48 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 06:32.13 | hackish | tkd, my question arrives after reading said instructions. |
| 06:32.27 | hackish | http://www.voip-info.org/wiki/view/Monitor+setup+sample |
| 06:33.00 | ChannelZ | Read these instructions: core show application Monitor |
| 06:33.05 | [TK]D-Fender | hackas I told you, the ones you should be reading are from * CLI |
| 06:33.09 | [TK]D-Fender | hackish: ^ |
| 06:33.28 | [TK]D-Fender | hackish: And that wiki is a wasteland and has been for some time |
| 06:34.00 | [TK]D-Fender | hackish: It is somewhere around the fifth-sixth place you should be looking for anything |
| 06:34.01 | ChannelZ | A teenage wasteland |
| 06:37.10 | hackish | yep, wrong docs suck |
| 06:37.41 | hackish | apparently SetVar doesn't work either... SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) |
| 06:37.48 | hackish | ideas on what I should be looking for? |
| 06:38.33 | ChannelZ | Set |
| 06:40.25 | [TK]D-Fender | hackish: SetVar died in Asterisk 1.2 .. in ***2005*** |
| 06:41.26 | [TK]D-Fender | hackish: VoIP-Info.org is the 3rd village they left out after Sodom & Gomorrah |
| 06:41.33 | hackish | lots of bitrot in documents. I frequently will google a problem to find a forum post or whatever so I know where to look. In these cases the voip-info site scores high with results but low with accuracy |
| 06:41.45 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 06:41.50 | *** part/#asterisk mihamina (~mihamina@41.188.46.121) |
| 06:42.51 | ChannelZ | do 'core show applications' and then go from there |
| 06:45.02 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 06:46.07 | hackish | pure ugly: Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}) |
| 06:47.25 | [TK]D-Fender | hackish: Nope... that's actually pretty sane |
| 06:47.26 | edong23 | yeah voipinfo.org is shart these days |
| 06:47.40 | edong23 | every now and then you can find a diamond in there, but mostly it is obsolete |
| 06:48.10 | *** join/#asterisk deo_ (~deo@122.53.72.218) |
| 06:48.17 | hackish | I wonder if it is not better to use mixmonitor |
| 06:48.29 | ChannelZ | Usually yes. |
| 06:48.35 | [TK]D-Fender | it takes a LOT of shit and a lot of pressure to make a diamond from that place... |
| 06:49.02 | hackish | I supose one could get really fancy and try to normalize each side of a conversation, then mix it, then transcode to MP3 or something. |
| 06:49.29 | edong23 | [TK]D-Fender: fair enough, diamond was being very generous... |
| 06:50.12 | hackish | however that brings up an interesting point... If you have a 30 minute conversation with someone and then it takes 2 minutes to transcode and archive it is that extension blocked during that time? |
| 06:50.32 | [TK]D-Fender | hackish: Depends how you do it |
| 06:50.59 | [TK]D-Fender | hackish: And why you'd by a CPU it'd take that long on... ick |
| 06:51.35 | hackish | personally I'd probably just MV the files somewhere and have a cronjob pick them up and code them. |
| 06:52.43 | [TK]D-Fender | hackish: run the cleanup in a BACKGROUND TASK |
| 06:53.06 | [TK]D-Fender | hackish: Which you can do and let the dialplan continue on its merry way if it doesn't need to wait |
| 06:53.12 | hackish | fender I'm going to assume 30 minutes of audio is a lot of data. I really don't know how long it would take. The only thing I transcode is video footage and I know it's slow in Vegas Studio. |
| 06:53.26 | ChannelZ | depends on how shit your server is |
| 06:53.36 | ChannelZ | and video is quite a different beast than audio. |
| 06:53.37 | hackish | ok, that does answer my question. |
| 06:53.45 | ChannelZ | Especially if it's only 8khz |
| 06:53.55 | [TK]D-Fender | in MONO |
| 06:54.02 | ChannelZ | It should take seconds, if that. |
| 06:54.06 | hackish | ok |
| 06:54.07 | [TK]D-Fender | 16 bit |
| 06:54.13 | [TK]D-Fender | up-hill |
| 06:54.15 | [TK]D-Fender | BOTH WAYS |
| 06:54.26 | [TK]D-Fender | in snow 20' high |
| 06:54.42 | hackish | most of my time is spent working in assembly with embedded processors. It's easy to sometimes lose focus of how quick real processors work. |
| 06:54.57 | hackish | looks out the window at a snowbank.... check |
| 06:57.33 | edong23 | hackish: should be nearly realtime honestly |
| 06:59.21 | hackish | got an mp3 encoder you recommend? |
| 06:59.33 | hackish | I seem to think ffmpeg is on this machine. |
| 07:00.12 | edong23 | that will work |
| 07:00.29 | edong23 | i think i still use lame |
| 07:00.47 | edong23 | that stuff is all automated and hasnt been touched in years on my servers... |
| 07:00.58 | edong23 | pretty sure its lame |
| 07:02.15 | hackish | looks like ffmpeg uses lame |
| 07:02.29 | hackish | rather than build all zillion lines of it I'll just use lame... |
| 07:03.16 | edong23 | yeah, ffmpeg is great for video, cause of that modularity |
| 07:03.29 | edong23 | hackish: why do you want to encode it though? |
| 07:03.39 | edong23 | the only reason i do is to email someone a voicemail |
| 07:03.46 | hackish | size |
| 07:04.31 | [TK]D-Fender | My phone plays back the default wav's just fine.. no need for MP3 here |
| 07:04.42 | [TK]D-Fender | and it would only make it bigger really... |
| 07:05.03 | hackish | ok I would have assumed that mp3 would be smaller than most .wav formats. |
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| 07:05.49 | edong23 | it never made it "bigger" per say, i never looked at the last bits in the size |
| 07:06.28 | edong23 | the wav files played fine for me, but other users couldnt do it on i think mac? |
| 07:06.36 | edong23 | for whatever reason |
| 07:06.39 | hackish | don't really play in audio, I just remember back in the day when mp3's were first invented you'd spend an hour and bring a 50mb wav file down to 1mb |
| 07:07.00 | edong23 | hackish: but that is a full fledged song (most likely) |
| 07:07.08 | hackish | yep |
| 07:07.09 | edong23 | this is 8khz MONO only |
| 07:07.19 | edong23 | its pretty minute |
| 07:08.03 | hackish | ok, I was told that voip would consume about 200k/sec. So I was figuring a 10 minute long clip will get hefty. |
| 07:08.25 | hackish | of course who knows how much packet overhead that includes |
| 07:08.50 | edong23 | well, math knows |
| 07:09.17 | edong23 | ulaw is an average of 80k/s though |
| 07:09.22 | [TK]D-Fender | RTP overhead does not exist in an audio file |
| 07:09.26 | edong23 | i dont remembe ractually |
| 07:11.16 | hackish | that doesn't look right... 142124 test.wav 9072 test.mp3 |
| 07:11.55 | edong23 | a 30minute call was 142k? |
| 07:12.03 | edong23 | is that k already? |
| 07:12.08 | edong23 | so 142M? |
| 07:12.09 | hackish | no that was a 5 second test |
| 07:12.18 | Cutzmf | Hi! Does anyone knows what permission asterisk used to open(create i mean) /var/spool/faxes/...? |
| 07:12.48 | edong23 | Cutzmf: what users is asterisk running as? |
| 07:13.11 | Cutzmf | aterisk.asterisk |
| 07:13.21 | edong23 | i would guess it is likely that |
| 07:13.27 | edong23 | <[TK]D-Fender> My phone plays back the default wav's just fine.. no need for MP3 here |
| 07:13.44 | edong23 | just saw that |
| 07:14.02 | Cutzmf | edong23: i mean creation permissons of file |
| 07:14.12 | edong23 | but yeah, i meant to email them the wav didnt work on apple devices (or at least some) |
| 07:14.22 | edong23 | oh, i see |
| 07:14.38 | [TK]D-Fender | edong23: I use VLC on my 3-year old Android |
| 07:14.39 | hackish | that 9k mp3 file is perfectly usable. |
| 07:14.40 | edong23 | Cutzmf: do you have a file it created? |
| 07:15.06 | edong23 | hackish: no doubt, it shoudl be |
| 07:16.19 | edong23 | [TK]D-Fender: i recently saw tht there was vlc for android, havent messed with it |
| 07:16.24 | edong23 | my phone sucks anyway |
| 07:16.29 | edong23 | i probably dont have room to install it |
| 07:17.25 | edong23 | verizon has utilized all my built in resources for some NFL shit and some radio thing.. and about 10 other giant applications that i will never use and cannot uninstall |
| 07:17.35 | [TK]D-Fender | I'm on a stock Samsung Galaxy S (v1) that's messed up and I can't upgrade the firmware on... |
| 07:18.03 | Cutzmf | edong23: in fact files created with 750, i need permissions to be 770 or 660 |
| 07:18.16 | [TK]D-Fender | Well ... at least with Samsung's "tool" |
| 07:19.09 | edong23 | hm... [TK]D-Fender might be able to help you there Cutzmf |
| 07:19.20 | edong23 | i havent tried to make asterisk change its file creation masks |
| 07:19.38 | [TK]D-Fender | No, I can't |
| 07:19.48 | [TK]D-Fender | Careful who you "volunteer" for help |
| 07:19.58 | Cutzmf | edong23: i need asterisk group open dirs/files with RW under /var/spool/asterisk/fax |
| 07:20.05 | hackish | time for sleep again... config is pretty much done.. thanks all for the help. |
| 07:20.15 | edong23 | well, i said might, removing obligation |
| 07:20.26 | edong23 | but Cutzmf you can do it ugly with a cron job |
| 07:21.50 | Cutzmf | yep... but in a time asterisk changes permissions to 750 for whole /var/spool/asterisk/fax and scp can't copy till crontab chmod is done with -R 760 |
| 07:23.23 | Cutzmf | so remote host needs logic to wait till fax dir is chmod'ed |
| 07:23.31 | edong23 | Cutzmf: http://www.asteriskguru.com/tutorials/asterisk_conf.html |
| 07:23.34 | edong23 | thats a quick google |
| 07:23.44 | edong23 | it does have some stuff about creation masks |
| 07:23.48 | edong23 | but i didnt read it much |
| 07:23.50 | edong23 | im goign to bed |
| 07:25.02 | edong23 | Cutzmf: is 750 what is set in your asterisk.conf? |
| 07:25.58 | WIMPy | Hi edong23. Got your callerid fixed? |
| 07:26.15 | edong23 | WIMPy: um.. i never had callerid issues.. |
| 07:26.41 | WIMPy | Wasn't that you with the TON thing? |
| 07:27.14 | edong23 | TON != callerid |
| 07:27.20 | Cutzmf | edong23: i have astctlpermissions = 0775 |
| 07:27.27 | WIMPy | Part of it. |
| 07:27.37 | edong23 | i did have a TON thing going on i havent gotten to testing my options |
| 07:27.53 | edong23 | i have 2 options, but no way to test while at home, and i have been sick this weekend |
| 07:27.55 | edong23 | just now feeling better |
| 07:27.59 | [TK]D-Fender | way past my bed-time, I'm off... |
| 07:29.09 | edong23 | me too, im out |
| 07:29.15 | edong23 | WIMPy: ill let you know tomorrow |
| 07:30.02 | Cutzmf | edong23: monitor uses 660, fax 750, i'm lost |
| 07:30.29 | edong23 | Cutzmf: no idea dude... i havent done it... |
| 07:30.41 | edong23 | but im pretty sure thats wher eyou would need to set it |
| 07:30.53 | Cutzmf | edong23: tnx! |
| 07:31.31 | edong23 | Cutzmf: you could possibly use inotify to only change new file permissions |
| 07:31.48 | edong23 | that would be faster, though hackish |
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| 07:36.00 | Cutzmf | edong23: yep it's useful with a litte bit of files? but if we have a big amount - inotify triggers use up to 1-3 min to setup |
| 07:37.41 | edong23 | inotify triggers take 1-3 minutes? |
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| 07:39.08 | Cutzmf | i've tryed with monitor with big dir tree with Gigs of files in dirs |
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| 07:43.40 | edong23 | Cutzmf: why are you keeping files in there? |
| 07:46.49 | Cutzmf | we have complex proprietary product with logig of file quota, but we have our fax solution which copies files to /var/spool/asterisk/fax |
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| 07:48.30 | Cutzmf | edong23: i mean that our company bought that product |
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| 08:48.02 | Chainsaw | drmessano: You're still managing to crash? That's interesting. |
| 08:48.15 | Chainsaw | drmessano: Did the RC help or is it worse then that? |
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| 08:53.35 | Rico29 | hi all |
| 08:54.27 | Rico29 | In asterisk 1.8, do Y have to set "envelope=yes" for voicemail boxes where I want to play the date the message was left ? (realtime voicemail) |
| 08:56.54 | ChannelZ | Well in non-realtime envelope is on by default, I'd figure that applies to RT as well but I couldn't say for sure. |
| 08:59.35 | Rico29 | ChannelZ> doesn't seem to work |
| 09:00.29 | Rico29 | http://pastebin.com/xA8d7F3z |
| 09:00.42 | Rico29 | when I leave a message, I have "sayCID", but no envelope |
| 09:14.01 | ChannelZ | hmm |
| 09:15.23 | ChannelZ | get any warnings or notices on the console? If your hit 3 and then 3 again on the message does it read the envelope? |
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| 09:18.10 | Rico29 | ChannelZ> no warnings |
| 09:18.25 | Rico29 | and if I hit 3,3, it only says callerid |
| 09:18.30 | Rico29 | maybe a tz problem ? |
| 09:25.06 | ChannelZ | yeah, that's a little weird |
| 09:28.57 | Rico29 | whyhow can I get more debug vrom app_voicemail ? |
| 09:29.10 | Rico29 | I've tried "core set debug 90 app_voicemail.c" |
| 09:29.16 | Rico29 | but nothing more on the cli |
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| 09:31.41 | ChannelZ | remove the .c so it's just app_voicemail |
| 09:34.04 | Rico29 | changes nothing |
| 09:34.57 | ChannelZ | do you have debug turned on for the console in logger.conf? |
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| 09:36.19 | Rico29 | of course |
| 09:37.25 | ChannelZ | well dunno why you're not seeing anything then |
| 09:38.03 | ChannelZ | is your vm storage setup in a database as well? |
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| 09:43.30 | Rico29 | ChannelZ> config in DB, messages in /var/spool/asterisk/voicemail |
| 09:43.36 | Rico29 | it plays well the CID |
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| 09:44.56 | ChannelZ | look at the msgXXXX.txt for the message in question, does it have an origtime? anything look odd? |
| 09:45.19 | ChannelZ | (pastebin one perhaps) |
| 09:45.46 | Rico29 | http://pastebin.com/vQWGTBgr |
| 09:46.15 | Rico29 | nothing looks strange |
| 09:46.51 | Rico29 | ChannelZ> found my problem for the debug, but no warnings in debug too, like if "envelope=yes" instruction was not set |
| 09:48.00 | ChannelZ | well if you "voicemail reload" with debug on you should see it say some things about global options |
| 09:48.40 | Rico29 | ok |
| 09:48.43 | Rico29 | let's try it |
| 09:50.26 | ChannelZ | but even so advanced options shouldn't care whether it's turned on or off. Glancing at the code, the place I see it returning early with no debug emitted is if it can't get the origtime from the message. |
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| 09:54.29 | Rico29 | ok |
| 09:56.23 | Rico29 | do I have to set "locale" in voicemail.conf ? in addition of tz |
| 09:57.34 | ChannelZ | I don't think that matters in this case |
| 09:58.07 | Rico29 | ok |
| 09:58.23 | Rico29 | mmmmmh |
| 09:58.29 | Rico29 | out of ideas |
| 09:58.32 | ChannelZ | is your tz something weird? |
| 10:00.04 | Rico29 | [general] |
| 10:00.08 | Rico29 | tz=european |
| 10:00.20 | Rico29 | [zonemessages] |
| 10:00.20 | Rico29 | european=Europe/Paris|'vm-received' Q 'digits/at' R |
| 10:03.25 | ChannelZ | not sure what else to say short of adding some more debug into the code and recompiling to see why it's not doing anything |
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| 10:32.15 | angryuser | Good day, i have an asterisk 10.3.1 on site, sometime i have some power outages, i can not fix them, the problem is when asterisk starts, for whatever reason it does not read the host of the peer [provider] in sip show peers i have Host : Unknown, however when i do a sip reload, it reads it. It is not even a domain name, how can i avoid doing sip reload for every power outage ? |
| 10:40.19 | apb1963_ | Get a UPS? |
| 10:40.59 | Chainsaw | angryuser: Consider upgrading to the 11 branch. |
| 10:41.28 | Chainsaw | angryuser: The 10 branch was always envisaged to be temporary, and there is no longer any development happening on it. 10->11 is effortless, config-wise. |
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| 10:46.43 | angryuser | apb1963_: sure 15 KVA, 20k |
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| 10:59.06 | Izeau | Hi, I have trouble connecting a websocket client (using JsSip) and a softphone (Blink) on Asterisk 11.2. |
| 10:59.15 | Izeau | 11.2.1* |
| 10:59.31 | Izeau | Both connect fine and show up in “sip show peers”, but cannot communicate |
| 11:00.21 | Izeau | The exact error message is: “ERROR[7101]: tcptls.c:446 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.191.173:52119: Connection refused” |
| 11:00.57 | Izeau | When the websocket client tries to call the Blink softphone |
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| 11:02.12 | Izeau | sip.conf: http://paste.awesom.eu/3Xl |
| 11:02.52 | Izeau | extensions.conf: http://paste.awesom.eu/OJH |
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| 12:05.14 | R1ck | soo.. I'm looking for a nice music on hold.. I'd preferable stream some radio station or something, does anybody have any ideas about this |
| 12:05.17 | R1ck | ? |
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| 12:32.49 | skrusty | R1ck: try something like shoutcast? use mpg123 |
| 12:33.25 | skrusty | you can pass mpg123 a url to stream from |
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| 12:50.46 | ectospasm | be wary of using an MP3 stream or file... they're usually recorded at 44.1kHz 16bit stereo, whereas most phones can still only play 8kHz 8bit mono... |
| 12:51.05 | ectospasm | there are "HD" phones out there that will sound better |
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| 13:21.27 | anonymouz666 | I am in trouble to analyze a beep frequency from a recording... moving from time domain to frequency domain |
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| 13:48.44 | R1ck | can I configure music on hold to stop playing after I hit the pound key? |
| 13:54.01 | leifmadsen | R1ck: yes, set it up so you can cycle through music on hold, and cycle to a silent moh group |
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| 13:54.51 | leifmadsen | look at the "digit" option in musiconhold.conf |
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| 13:55.45 | R1ck | ahh |
| 13:56.08 | leifmadsen | I usually just create a [silent] section that uses /var/lib/asterisk/sounds/en/silence/ as the source |
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| 14:09.46 | skrusty | not sure if anyone is interested, but i've put a small call pcap utility i wrote in c# up on codeplex. It lets you dump sip calls to individual pcap files. http://sipdump.codeplex.com/ |
| 14:10.31 | barbosa2 | hi for all, is possible send SMS using AEX800 ? |
| 14:11.34 | [TK]D-Fender | barbosa2, No |
| 14:12.07 | barbosa2 | [TK]D-Fender, thanxx |
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| 14:21.24 | R1ck | hmm, those digit= things don't work, should I somehow enable the feature? |
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| 14:30.47 | [TK]D-Fender | R1ck, What version are you using? |
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| 14:31.18 | leifmadsen | R1ck: what version of asterisk? |
| 14:31.48 | leifmadsen | should "just work", there is nothing to enable other than to make them part of the configuration and the module be reloaded |
| 14:32.04 | leifmadsen | suppose it might be possible you have to unload then load res_musiconhold.so |
| 14:32.33 | R1ck | Asterisk 1.8.10.1~dfsg-1ubuntu1 |
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| 14:32.45 | leifmadsen | should work fine, the example I saw was in 1.8 |
| 14:33.15 | R1ck | hm nope. is it because this is in a Queue perhaps? |
| 14:33.25 | leifmadsen | probably |
| 14:33.29 | [TK]D-Fender | R1ck, pastebin your MoH config... and you did restart * completely, right? |
| 14:33.31 | leifmadsen | the queue() is likely eating the dtmf |
| 14:33.43 | [TK]D-Fender | r1and yes, this would be because of being in a queue |
| 14:34.01 | R1ck | hmm |
| 14:34.01 | [TK]D-Fender | R1ck, that uses DTMF with "context=" for an ext option |
| 14:34.04 | [TK]D-Fender | exit* |
| 14:34.19 | R1ck | I see |
| 14:35.09 | R1ck | is there no way, in a Queue, to let users cycle between MoH classes? I figured it'd be cool to have them select a stream of a radio station :) |
| 14:36.48 | [TK]D-Fender | R1ck, You have the source code..... |
| 14:37.03 | R1ck | ugh :) |
| 14:37.30 | leifmadsen | you'll have to look at the options available to the queue to see if that is possibl;e |
| 14:37.38 | leifmadsen | core show application queue, and look at queues.conf |
| 14:40.32 | anonymouz666 | very hard to detect a beep frequency with asterisk |
| 14:40.44 | R1ck | well basically why I wanted it in the first place, was to give users the ability to cancel the MoH alltogether. I can see with option 'r' it can ring instead of playing MoH, but I guess its not possible to have a caller change that option? |
| 14:41.32 | leifmadsen | I understand what you're trying to do. I'm saying you'll need to look through the options and configuration to determine if such a thing is possible. It may not be through the Queue() application. |
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| 15:01.51 | nunne | Running 11.2.1 and trying to use res_calendar for iCal.. i Get "Unable to parse iCalendar xx from xxxxx: 302 Redirect". Anyone know a fix around this error? |
| 15:03.21 | Chainsaw | nunne: 302 is a HTTP code; it would suggest that your iCAL URL is not correct. |
| 15:03.39 | Chainsaw | nunne: Try to wget from that URL, and see what it redirects you to. Then use the ultimate link target instead. |
| 15:05.22 | nunne | Chainsaw: ofcourse.. thanks! dunno why i didn't think of that :P |
| 15:06.01 | Chainsaw | nunne: Not to worry. I like easy questions. They make me look good for knowing the answer ;) |
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| 15:33.19 | Phoebus | Hello folks, anyone up to partneting up for a consulting firm, voip dept? |
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| 15:33.29 | Phoebus | Still in the early stages, just the right time to get in. |
| 15:33.39 | Phoebus | We work online, so I don't care where you live. |
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| 15:53.10 | igcewieling | This is SO annoying. Only allowing g729, the g729 sound files are installed, but Asterisk is still playing the .ulaw version of the sound files. |
| 15:53.31 | igcewieling | yes, there are g729 licenses |
| 15:53.37 | Greenlight | That's very odd |
| 15:53.47 | file | and format_g729.so is loaded? |
| 15:57.25 | Yxa | hi all. currently I have my asterisk server in my lan (192.168.1.4). I want to let remote clients connect but I do not want to do a direct NAT as that would open my local network to the public. So I'm thinking of putting another asterisk in the DMZ and let them handle remote clients. Can that be done? |
| 15:57.39 | igcewieling | file: you are a genius. any reason it would not be loaded? there is no noload line for it in modules.conf |
| 15:57.48 | file | no idea |
| 15:58.04 | Greenlight | Check the CLI output when it's starting |
| 15:58.08 | igcewieling | maybe I didn't restart asterisk after loading the codec_g729. In any case, thanks! |
| 15:58.25 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-147-150.user.veloxzone.com.br) |
| 15:58.40 | Greenlight | Yxa: Yes |
| 15:59.22 | Yxa | Greenlight how do i sync the accounts on both machines? Or is there such a thing as pure forwarding mode? |
| 16:00.07 | Greenlight | No such thing as forwarding mode |
| 16:00.21 | Greenlight | How you sync the accounts is up to you. |
| 16:00.42 | Greenlight | It does seem overkill though. Why not just foward the neccissary ports to the main asterisk box? |
| 16:00.53 | Greenlight | Or, give it a public IP |
| 16:01.00 | [TK]D-Fender | Yxa, * is not a SIP proxy |
| 16:01.41 | Yxa | Greenlight i want to avoid that. would rather a box in dmz gets compromised |
| 16:02.10 | Greenlight | Yea, but if the box in the DMZ gets "compromised" then are you not going to have that box accessing the main box anyway ? |
| 16:02.12 | Yxa | [TK]D-Fender yeah. any suggestions? |
| 16:02.15 | [TK]D-Fender | Yxa, You are opening SIP either way. |
| 16:02.18 | Greenlight | Indeed |
| 16:02.20 | Greenlight | ^^ |
| 16:02.27 | [TK]D-Fender | Yxa, What makes yout hink the level of expose is any different? |
| 16:02.35 | [TK]D-Fender | exposure* |
| 16:02.46 | Greenlight | Your adding complexity, and gaining nothing |
| 16:02.52 | Greenlight | *You're |
| 16:05.28 | Yxa | hmm the main box will have direct access to PRI. was thinking adding another box would at least put another layer of security |
| 16:05.37 | jmetro | in the end, just stick norton on i. |
| 16:05.59 | Greenlight | Yxa: I must have misunderstood. Where is the extra layer of security ? |
| 16:06.20 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:06.21 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:06.51 | Yxa | Greenlight 'was thinking' :) |
| 16:07.04 | Greenlight | Ahh |
| 16:07.19 | edong23 | Yxa: isnt that what an SBC is for? |
| 16:07.53 | Greenlight | If someone got into the 2nd box, they'd have full access to PRI via the first anyways |
| 16:07.57 | [TK]D-Fender | In the end the SIP packets are still going to make it to your server just the same. |
| 16:08.18 | [TK]D-Fender | if that device has rights to dial out it doesn't matter how many intermediaries it goes through |
| 16:08.21 | [TK]D-Fender | auth is auth |
| 16:08.39 | Greenlight | Depending on the nature of your setup, you could use a VPN, or IP-based restrictions to secure stuff down a litte more |
| 16:08.58 | Greenlight | You might also restrict calls to specific destinations etc just in case |
| 16:09.41 | Yxa | yeah I guess it makes sense |
| 16:09.47 | Greenlight | For our "public" facing SIP stuff, we restrict to UK-based IP's and use fail2ban to block the IP after 3 attempts. So far in last 18 months we've not had any hacking attempts from UK IP's though |
| 16:10.10 | Greenlight | Before the UK only restruction, we had a number of attacks per day |
| 16:10.15 | edong23 | Greenlight: same here -ish... not 3 attempts |
| 16:10.32 | Greenlight | Yea, with fail2ban the attempts thing is not exact anyways |
| 16:10.34 | Yxa | Greenlight what kinda attacks? sip registration? |
| 16:10.35 | Greenlight | Since it polls the logs |
| 16:10.49 | Greenlight | Yxa: Yea, people trying their luck I guess |
| 16:10.53 | edong23 | combine that with UUID based passwords that are long as hell, and you have pretty good security |
| 16:11.30 | edong23 | the last "admin" that worked in my office had sip user <exten> and password <exten> |
| 16:11.31 | edong23 | lol |
| 16:11.36 | Greenlight | lol |
| 16:11.37 | edong23 | it got busted pretty quick |
| 16:11.50 | edong23 | and it took him an entire day to figure it out |
| 16:11.53 | Greenlight | Yea, it's amazing how quickly thigs get compromised with poor security |
| 16:12.01 | edong23 | yeah |
| 16:12.08 | edong23 | asterisk itself doesnt need an SBC |
| 16:12.12 | Greenlight | remembers £3k worth of calls to Gaza Strip |
| 16:12.17 | edong23 | but im trying to build an sbc for our other gear |
| 16:12.27 | Greenlight | We reckoned local authorities wouldn't care ... |
| 16:12.30 | edong23 | the class 5 switch doesnt have the same capabilities |
| 16:13.10 | edong23 | heh, that attack was saved by the fact that he only ran the trixbox (yeah, i know) |
| 16:13.18 | edong23 | i controlled the pri and the sip terminations |
| 16:13.24 | edong23 | so they couldnt call outside our country |
| 16:13.45 | edong23 | but they hammered out calls about car warranties and all kinds of craziness fraud |
| 16:13.52 | Greenlight | heh |
| 16:13.53 | edong23 | i let it go for 1 entire day too |
| 16:13.57 | edong23 | so he could figure it out |
| 16:14.32 | edong23 | this is the same guy that put QoS rules for port 5060 |
| 16:14.34 | Greenlight | Yea, restricting channel counts and destinations is wise. We do it on our billing gateway, so each call is autherised by a stored proc for that particular customer, against a table of destinations |
| 16:14.37 | edong23 | i was like "fantastic!!!" |
| 16:14.47 | Greenlight | Wow.... QoS that control traffic lol |
| 16:15.04 | Greenlight | WE can't have an 180 RINGING gettign delayed! |
| 16:15.12 | edong23 | yeah |
| 16:15.38 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:16.09 | edong23 | when it didnt do anything for the call quality on our wireless netowkr, i just laughed because he said the gear had cpus that just "couldnt keep up" |
| 16:16.17 | edong23 | i was like "you havent touched rtp" |
| 16:16.18 | *** join/#asterisk phiscribe (~david@98.157.5.116) |
| 16:16.22 | Greenlight | heh |
| 16:16.53 | edong23 | he wasnt here much longer after that |
| 16:18.06 | Yxa | edong23 i am trying to read up on SBC. it seems to be a device to put between asterisk and remote clients. But you mention asterisk doesnt need that. Could you elaborate? |
| 16:18.56 | carrar | Go purchase a Sonus SBC :) |
| 16:19.04 | carrar | two actually |
| 16:19.10 | carrar | wanna be redundant |
| 16:21.14 | Yxa | carrar what kinda price range am I looking at? |
| 16:21.17 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 16:21.45 | carrar | You're looking at the top end of the price range |
| 16:21.55 | [TK]D-Fender | Yxa, if the same packets are making it end-to-end.. then your exposure is the same |
| 16:22.05 | Greenlight | You don't need an SBC |
| 16:22.07 | [TK]D-Fender | Yxa, I haven't heard any functional difference... |
| 16:22.24 | Greenlight | It's a way for Cisco and the likes to bend you over even more and take your cash |
| 16:22.34 | [TK]D-Fender | Yxa, The same people can hammer the same accounts the same way and end up with the same results.... |
| 16:22.56 | WIMPy | But it feels safer. |
| 16:23.01 | carrar | Yxa, why not put your sip server behind a FW and mae people use a VPN? |
| 16:23.07 | carrar | make |
| 16:24.32 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 16:24.32 | *** mode/#asterisk [+o file] by ChanServ |
| 16:24.39 | leifmadsen | file: masochist. |
| 16:25.04 | Yxa | carrar i could for pcs and softphones. but what about hardphones |
| 16:25.20 | carrar | what about them? |
| 16:25.30 | Greenlight | Use the VPN from the router at the remote side |
| 16:25.42 | Greenlight | Some hardphones even allow VPN connection directly |
| 16:25.58 | Greenlight | Or... if the remote side can get a fixed IP even better |
| 16:26.02 | Greenlight | Lots of ways to secure things |
| 16:26.14 | Greenlight | Very rarely do you need to just open SIP to the world |
| 16:26.30 | Yxa | Greenlight does sip send the password in cleartext during registration? |
| 16:27.08 | carrar | lets see |
| 16:27.23 | Greenlight | Yxa: I beleive so yes |
| 16:27.44 | file | erm, no |
| 16:27.55 | Greenlight | Maybe not then ... :P |
| 16:28.09 | leifmadsen | no, it does not |
| 16:28.09 | file | authentication is digest challenge, so while the response to the challenge is in cleartext (if using UDP or TCP) the contents are not the password in cleartext |
| 16:28.11 | leifmadsen | it uses a hash |
| 16:28.19 | file | it's hashed a few times with different bits of data |
| 16:28.25 | leifmadsen | that ^^ |
| 16:28.26 | Greenlight | Ahh |
| 16:31.21 | Yxa | at least that sounds alittle more comforting |
| 16:33.10 | edong23 | Yxa: i had the same nay-sayer attitude before... but its not as black and white as that |
| 16:33.23 | edong23 | and sbc is needed when you cant fully control and trust the device |
| 16:33.33 | edong23 | asterisk isnt like that, you can put your security on it |
| 16:34.04 | edong23 | so you would have a meek set of features come out of an sbc for your setup |
| 16:34.21 | Yxa | edong23 i hear you |
| 16:34.40 | edong23 | but if you didnt have that kind of control, you would likely want an sbc |
| 16:35.10 | edong23 | for instance, a device talking sip to the world that you cannot even really log into to find out if you are getting hammered with auth attempts |
| 16:35.43 | edong23 | there are a few other reasons for an sbc but as carrar was (trollingly) hinting to, an sbc is a huge cost |
| 16:35.46 | edong23 | none of them are cheap |
| 16:35.57 | carrar | WHA :) |
| 16:35.58 | edong23 | im currently looking into YATE as a possible free sbc |
| 16:36.24 | edong23 | before my company spends 30k on a single (no redundancy) sbc |
| 16:36.37 | edong23 | yate claims it can be an sbc, but i dont know about its feature set |
| 16:36.48 | edong23 | its sip stack is for sure better than the default asterisk sip stack |
| 16:36.57 | Yxa | isn't there an opensbc or something? |
| 16:37.09 | edong23 | opensips can also be an sbc |
| 16:37.19 | edong23 | Yxa: no... i mean |
| 16:37.32 | edong23 | tbh and sbc isnt like a set number of things that can make it that simple |
| 16:38.00 | edong23 | they offer signalling control, nat transversal, monitoring capability, and dos protection |
| 16:38.01 | file | Asterisk 12 will feature a brand new SIP channel driver with a third party SIP stack! Dun, dun, dunnnnn. |
| 16:38.17 | edong23 | those things can be done on any number of devices... |
| 16:38.38 | edong23 | Yxa: if you can wait a few days, ill have my test sbc up and the quotes from 13 vendors on sbc prices |
| 16:39.08 | edong23 | file: havnt you been able to incorporate 3rd party sip stacks in asterisk for a while? |
| 16:39.18 | edong23 | its only been that the default sip in asterisk lacks some functionality |
| 16:39.31 | edong23 | i have never had an issue with it, as i dont need sip proxy and the likes |
| 16:39.34 | file | you can always plug new channel drivers in |
| 16:39.39 | edong23 | yeah |
| 16:39.45 | edong23 | someoen told me they did before |
| 16:39.52 | file | but to really replace chan_sip is not something you can just do in a week |
| 16:40.02 | edong23 | but asterisk is swiss army knife as it is, i dont want to touch it |
| 16:40.08 | edong23 | its great |
| 16:40.09 | Yxa | edong23 how do you justify spending 30k on a machine with PRI that prob costs max 5k |
| 16:40.19 | edong23 | Yxa: you dont |
| 16:40.25 | edong23 | but there are other options |
| 16:40.31 | edong23 | thast just where i told my company to wait |
| 16:40.38 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 16:40.52 | edong23 | there were literally signing contracts with acme packet |
| 16:41.17 | edong23 | but (for instance) as far as service provider sbcs go squire technologies has a nice one that just installs on centos |
| 16:41.22 | edong23 | its much cheaper |
| 16:41.50 | edong23 | if you are on the enterprise side... then you would look at an enterprise solution, like avaya |
| 16:41.57 | edong23 | or audiocodes |
| 16:42.16 | edong23 | but they are a rip off as carrar was suggesting |
| 16:42.22 | edong23 | they charge for the number of sessions |
| 16:42.25 | edong23 | licensed |
| 16:42.58 | edong23 | no matter how powerful the hardware you install it on, you will be limited to that many sessions without upgrading your license |
| 16:43.09 | edong23 | its a joke, but a joke soem people have to be a part of |
| 16:43.17 | edong23 | by some people, i mean me for sure |
| 16:43.50 | carrar | Sonus is the gold standard in the carrier class |
| 16:44.24 | carrar | But I don't think Yxa is a global enterprise carrier |
| 16:44.37 | Yxa | no i'm not |
| 16:44.54 | carrar | Check out opensips |
| 16:45.01 | coppice | sonus is the biggest source of interoperability problems in the entire industry |
| 16:45.02 | carrar | freeswitch or yate |
| 16:45.27 | carrar | I've not had any issues with Interop & Sonus |
| 16:46.01 | carrar | (I don't run Sonus) |
| 16:46.05 | carrar | just interface with it |
| 16:46.37 | coppice | these days sonus tends to interoperate simply because other people have had to develop workarounds to tolerate their bugs |
| 16:46.48 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 16:47.35 | Yxa | harsh comments on a supposed market leader :) |
| 16:47.47 | edong23 | all market leaders get that |
| 16:48.03 | edong23 | but i think acme is the leader in sheer number of deployments |
| 16:48.32 | edong23 | oddly enough, the only thing that stopped us from buying acme was my argument that oracle bought them |
| 16:48.37 | edong23 | thank god for that |
| 16:49.09 | edong23 | but either way, it doesnt matter, ultimately an sbc should be standard. so any good sip stack should be able to partake as an sbc |
| 16:49.23 | carrar | Yxa, everything out there has it's bugs |
| 16:49.29 | edong23 | opensips was my first choice, but i have since moved to yate as a possibility |
| 16:49.39 | edong23 | fuck freeswitch and its xml configuration files |
| 16:49.40 | edong23 | i hate that |
| 16:49.55 | carrar | hahah |
| 16:50.04 | Yxa | edong23 there's a plugin for asterisk style conf |
| 16:50.12 | *** join/#asterisk Brixius (~quassel@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 16:51.30 | edong23 | a plugin to make asterisk xml? |
| 16:51.32 | edong23 | thanks but no thanks |
| 16:51.38 | edong23 | im not a machine, i hand edit |
| 16:53.03 | edong23 | Yxa: but as said, you shouldnt really need an sbc |
| 16:53.20 | edong23 | i have had asterisk installed for um... probably 7 years... 6 years? |
| 16:53.39 | edong23 | and ive only had a few issues that were fixed immediately |
| 16:53.52 | edong23 | just control your box like you would any other server |
| 16:54.03 | edong23 | and have good credentials |
| 16:54.11 | edong23 | and monitor your logs |
| 16:54.16 | edong23 | and be a good little sysadmin |
| 16:54.22 | edong23 | youll be fine without an sbc |
| 16:54.36 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 16:54.56 | Yxa | nods |
| 16:54.58 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:56.33 | *** join/#asterisk Brixius (~Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 16:56.42 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 16:56.45 | Yxa | so for my situation to allow remote phones, just port forward sip and rtp in and restrict ip ranges and dialplan? |
| 16:57.39 | edong23 | i woudl dmz it, and handle the firewalling on the host itself |
| 16:57.51 | edong23 | i kinda forgot about you saying its on a private address |
| 16:58.00 | Yxa | it's already on private |
| 16:58.04 | Yxa | can't move it |
| 16:58.09 | [TK]D-Fender | Port forward. No reason to expose anything more than that |
| 16:58.51 | Yxa | [TK]D-Fender 5060 and 10000-20000 is sufficient? |
| 17:01.04 | [TK]D-Fender | Yxa, Those are the sample ranges |
| 17:03.39 | Yxa | [TK]D-Fender that's what I use internally |
| 17:08.19 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:12.30 | jeffspeff | so, when my ram is used for caching, like when using scp to copy a file over, asterisk sees the cached ram as used. i have the minfreemem setting in asterisk.conf set to 2gb, but it doesn't take into account cached memory and so asterisk locks up |
| 17:14.17 | ChannelZ | caching in that way is really the domain of the OS |
| 17:15.38 | Qwell | Cached memory is used. |
| 17:16.32 | ChannelZ | and 2gb is kind of a lot |
| 17:16.45 | jeffspeff | the system has 16gb |
| 17:17.00 | jeffspeff | i'm using centos |
| 17:17.18 | jeffspeff | when i brought this up in that channel i was referred to this page http://www.linuxatemyram.com/ |
| 17:17.30 | Qwell | and did you read it? |
| 17:17.37 | jeffspeff | heres a pastebin of what my memory looked like http://pastebin.com/7RtCm7Ld |
| 17:18.01 | jeffspeff | Qwell, yes i read it, it swears up and down that cached memory is still available to applications and won't interfere with them |
| 17:18.18 | Qwell | It's available for allocations. |
| 17:18.22 | Qwell | It is, however, very much used. |
| 17:18.47 | Qwell | There's a handy table that explains it. |
| 17:19.04 | jeffspeff | so, how do we get asterisk to see that it can use it? so that asterisk will stop thinking [Feb 26 10:42:42] WARNING[3946][C-0000118b]: pbx.c:6536 increase_call_count: Available system memory (~258MB) is below the configured low watermark (2048MB) |
| 17:19.23 | Qwell | You don't. It's used. |
| 17:19.48 | Qwell | If you don't want that, then disable the checking. |
| 17:19.59 | jeffspeff | i did |
| 17:20.06 | Qwell | Then what's the problem? |
| 17:20.18 | jeffspeff | but if asterisk actually needs to the memory, will it grab it? |
| 17:22.13 | jeffspeff | i guess i'm not understanding why the minfreemem setting doesn't take cached mem into account. i thought that setting was there to make sure asterisk wouldn't use all the system memory; if asterisk knows it can use the cached memory then why does it freak out when some of the ram is cached but still available? |
| 17:22.32 | Qwell | Because they are different things. |
| 17:23.23 | Qwell | Applications have no idea why memory is used. If it asks how much is free, it is told. If it requests memory, it'll either get it or it won't. |
| 17:23.41 | jeffspeff | but cached mem is still avaible for apps to reclaim if needed, so there's no need for asterisk to throw errors that the system is almost out of ram |
| 17:23.42 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:23.47 | ChannelZ | minfree is just an arbitrary test, it's not related to its needs |
| 17:23.55 | Qwell | It *IS* out. |
| 17:24.04 | Qwell | Re-read that site about 5 more times. |
| 17:24.09 | jeffspeff | lol |
| 17:24.24 | jeffspeff | Linux is borrowing unused memory for disk caching. This makes it looks like you are low on memory, but you are not! Everything is fine! |
| 17:24.31 | jeffspeff | This makes it looks like you are low on memory |
| 17:24.42 | ChannelZ | Memory is meant to be used |
| 17:25.04 | jeffspeff | ok, so i guess the minfreemem setting is completely useless |
| 17:25.28 | jeffspeff | since it doesn't take into account cached mem that can be reclaimed by asterisk or any other app if needed |
| 17:25.30 | ChannelZ | Apps don't check to see how much RAM is free before they allocate it and fail if they don't "see" enough first, they attempt to allocate what they need and the OS either gives it to them or doesn't |
| 17:27.15 | jeffspeff | if the os will give cached mem back to applications upon request, then whatever command updates minfreemem needs to see that there is cached memory and not throw warnings right? |
| 17:27.31 | ChannelZ | However minfree DOES look, but it's not related to a specific need on the part of asterisk. It's an "oh shit" alarm |
| 17:28.54 | ChannelZ | just turn it off. It's clearly not doing what you think or want so turn it off. |
| 17:30.15 | drmessano | and then set up monit, pick a real metric, and alarm for it |
| 17:35.06 | *** join/#asterisk kresp0 (~screspo@80.100.216.87.dynamic.jazztel.es) |
| 17:46.00 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 17:46.25 | ulogic | does anybody want to discuss app_queue with me? |
| 17:47.43 | [TK]D-Fender | ulogic, Just ask your question |
| 17:48.16 | ulogic | There are a number of issues with using realtime with app_queue |
| 17:49.21 | ulogic | When using realtime, I can insert and delete from queue_members table and they are displayed when issuing "queue show" |
| 17:49.49 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:51.23 | ulogic | However, there are no manager events issued. If I call the AddQueueMember application, it fails because it is already added. I think there should be an option R for all the Queue applications to update Realtime |
| 17:51.54 | ulogic | Also when the queue is set for autopause, members in realtime do not get paused. |
| 17:52.32 | [TK]D-Fender | ulogic, What version are you using exactly? |
| 17:52.47 | leifmadsen | that's because queue_members are a separate realtime configuration |
| 17:53.04 | leifmadsen | pausing does work as I've done it |
| 17:53.20 | ulogic | I am using 11.3.0-rc1 |
| 17:53.30 | leifmadsen | you sure are. |
| 17:56.02 | ulogic | As I said, I can insert and delete in the queue_members table and they do get calls routed to them when there is a row present for them in the table, but I don't see anywhere in the code where it updates the status in realtime table in the function rna (Ring No Answer) |
| 17:57.51 | ulogic | I would also like to have manager events fire when adding or removing a member from the queue_members database table, but there is no mechanism to fire an ad hoc manager event from the dialplan. |
| 17:58.47 | ulogic | That's why I was thinking it would be nice to an "R" option for realtime to all of the applications supported by app_queue.c |
| 17:59.36 | kresp0 | ulogic: not sure, but maybe you should discuss that on #asterisk-dev |
| 18:00.13 | kresp0 | btw, hi all :) |
| 18:00.25 | ulogic | Thanks, I was unaware of the #asterisk-dev channel |
| 18:00.59 | kresp0 | you're welcome |
| 18:02.40 | kresp0 | I'm having trouble with calls that cannot be heard on one side... and I'm not using NAT. If I make a call from my mobile phone to my * via PSTN and enter the echo test, it works. However, if I do the same but calling to a SIP phone instead, the person on the SIP phone cannot hear me. And, echo test from that SIP phone works. |
| 18:02.50 | kresp0 | What should I try to debug that? |
| 18:03.34 | ChannelZ | You aren't using NAT, but is the other end? |
| 18:03.52 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:03.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:03.57 | *** join/#asterisk lorsungcu (~anonymous@75-146-164-137-Minnesota.hfc.comcastbusiness.net) |
| 18:04.19 | kresp0 | neither |
| 18:04.43 | kresp0 | mobile -> PSTN -> dahdi -> * -> SIP phone |
| 18:04.45 | ChannelZ | well I guess you could start with an rtp debug and see if it's even sending media to the right place first |
| 18:05.53 | kresp0 | thanks ChannelZ, I'll try that |
| 18:08.07 | kresp0 | got lots of this: |
| 18:08.07 | kresp0 | Sent RTP packet to 192.168.2.118:16468 (type 00, seq 002957, ts 138560, len 000160) |
| 18:08.08 | kresp0 | Got RTP packet from 192.168.2.118:16468 (type 00, seq 004196, ts 229727862, len 000160) |
| 18:08.25 | kresp0 | 2.118 is the SIP phone IP |
| 18:10.06 | ChannelZ | so they're talking to each other. I guess you need to look at the SIP phone its self then |
| 18:10.27 | ChannelZ | What codec? |
| 18:11.14 | ChannelZ | although it's curious that the echo test works from the device |
| 18:11.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:12.21 | kresp0 | ulaw |
| 18:12.23 | kresp0 | yep |
| 18:12.41 | [TK]D-Fender | Not really. |
| 18:13.49 | ChannelZ | well if it's a barf on the bridge between DAHDI and the SIP side |
| 18:19.22 | kresp0 | [TK]D-Fender, not really what? |
| 18:21.08 | [TK]D-Fender | Look at the complete call. |
| 18:21.12 | [TK]D-Fender | Always |
| 18:21.16 | [TK]D-Fender | show actual configs |
| 18:21.20 | [TK]D-Fender | This is not a guessing game |
| 18:23.46 | kresp0 | OK sorry. I'll pastebin the complete call and configs |
| 18:24.53 | *** join/#asterisk lorsungcu (~anonymous@75-146-164-137-Minnesota.hfc.comcastbusiness.net) |
| 18:24.57 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 18:25.10 | *** part/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 18:29.55 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:34.01 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 18:40.52 | kresp0 | sooo, this are my config files and call log: |
| 18:41.04 | kresp0 | call log -> http://pastebin.com/wfVa8Rcf |
| 18:41.04 | kresp0 | extensions.conf -> http://pastebin.com/aYJWXveN |
| 18:41.04 | kresp0 | macros.conf -> http://pastebin.com/en3rP6cK |
| 18:41.04 | kresp0 | errors.conf -> http://pastebin.com/iY9AWEki |
| 18:41.04 | kresp0 | sip.conf -> http://pastebin.com/iq6sZVAi |
| 18:41.04 | kresp0 | dahdi-channels.conf -> http://pastebin.com/ZejjGikG |
| 18:41.06 | kresp0 | chan_dahdi.conf -> http://pastebin.com/gDpkdJff |
| 18:42.33 | kresp0 | I mean: ...THESE are.. |
| 18:42.45 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 18:45.57 | *** join/#asterisk crazed1 (~themrrobe@unaffiliated/themrrobert) |
| 18:46.44 | hackish | When you call Goto, is the dialplan interpreter actually just replacing the context, extension and priority and adjusting the current instruction? So I could call goto to switch contexts and expect it's same=> instructions to still match the extension properly? |
| 18:47.22 | hackish | Or alteratively is there more of a function call type command I haven't found? |
| 18:47.46 | [TK]D-Fender | hackish, It goes exactly where you tell it |
| 18:48.04 | hackish | Fender, yes but I'm interested in the next instruction it runs. |
| 18:48.17 | [TK]D-Fender | hackish, Once you're there, "same" continues that pattern in that context just like normal |
| 18:48.18 | igcewieling | hackish: a goto never returns to to where it came from |
| 18:48.35 | igcewieling | if you want it to return to the next extensions.conf line use a gosub |
| 18:48.41 | [TK]D-Fender | hackish, GOSUB is used for Subroutines\ |
| 18:48.43 | Qwell | You mean like GoSub? |
| 18:49.12 | hackish | aha, that clears it up. |
| 18:49.25 | leifmadsen | *coughs* about asteriskdocs.org |
| 18:49.55 | hackish | I've just been reading all these examples that look like spaghetti code I wrote when I was in elementary school. |
| 18:50.01 | igcewieling | looks like hackish has learned enough to have his questions answered in the docs. |
| 18:50.28 | hackish | of course I just loaded and read the docs on it. Sometimes knowing what to search for is the biggest problem. |
| 18:50.43 | hackish | I was looking for func or call never thought about gosub |
| 18:50.43 | igcewieling | hackish: AEL is more "programming-like", but best used after you are familiar with extensions.conf syntax |
| 18:50.49 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 18:51.01 | igcewieling | hackish: "core show applications" and "core show functions" |
| 18:52.00 | McBoingBo | trying to take advantage of automon, reload res_features.so shows me "Remapping feature One Touch Monitor (automon) to sequence '*1'" but on calls *1 does not do anything, any ideas? |
| 18:56.10 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.159.79) |
| 18:56.40 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:56.42 | [TK]D-Fender | McBoingBo, Show us your full attempt |
| 19:03.18 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:06.15 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.159.79) |
| 19:08.46 | McBoingBo | [TK]D-Fender, put automon parameters in features.conf under [featuremap] "automon => *1", and added under [globals] DYNAMIC_FEATURES=automon. About to try and put Set(__DYNAMIC_FEATURES=automon) under my ext |
| 19:09.22 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 19:10.01 | McBoingBo | [TK]D-Fender, I mean, trying to use Set method because not getting any response/recording using *1 (reload res_features.so applied as well) |
| 19:10.33 | ChannelZ | does your Dial include the option to enable features? |
| 19:11.14 | McBoingBo | no, I imagine thats where my confusion lies, I see what I mentoined as 2 method, but it sounds like I need all the bits |
| 19:11.56 | ChannelZ | Look at w/W according to your desires |
| 19:12.39 | ChannelZ | although the dynamic features hmm |
| 19:14.06 | McBoingBo | ChannelZ, I used this "exten => 248,1,Set(__DYNAMIC_FEATURES=automon)" no workie either |
| 19:16.34 | jmetro | still wishing asterisk had a top-down strategy |
| 19:16.40 | ChannelZ | well isn't dynamic features meant for your own defined ones in the applicationmap? |
| 19:17.12 | McBoingBo | ChannelZ, I am just trying everything suggested/read |
| 19:17.14 | ChannelZ | automon is a 'normal' or built-in feature for lack of a better term |
| 19:17.27 | ChannelZ | so I don't think your example has any meaning |
| 19:18.05 | ChannelZ | so in this case if you're happy with automon being *1 you would want to add W or w to your Dial options |
| 19:18.14 | McBoingBo | ChannelZ, show features certainly thinks its enabled, and I get it now that I need to allow users to use the feature, tried exten => 248,1,Dial(SIP/248,20,Ww) |
| 19:18.27 | McBoingBo | yeah I just did that, nothing in console |
| 19:19.26 | fprior | Hi all, anyone can help me with this ? pastebin.com/0eXAp1Yr - rfc2833 arrives fro SIP trnuk but no dialplan reaction. Any idea ? |
| 19:20.50 | ChannelZ | works here, do you have verbosity on a little? |
| 19:21.14 | McBoingBo | ChannelZ, I see where my fault lies, I want to call OUT and turn on Monitor, but all these examples only work if someone calls IN |
| 19:21.35 | ChannelZ | fprior: I suppose that depends on your dialplan |
| 19:22.33 | ChannelZ | McBoingBo: there should be no difference.. it's just a matter of the Dial that initiates whatever bridge it is (whether from an incoming call TO a device, or a device making an outgonig one) needs to have the magic options |
| 19:23.03 | McBoingBo | ChannelZ, well I tested it and it only records on *1 when the call came into my exten |
| 19:23.10 | ChannelZ | and you have to think about the direction of the call as to which to use, W or w |
| 19:24.22 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 19:24.30 | ChannelZ | You pasted the Dial(SIP/248,20,Ww) example, but did you modify your 'outgoing' extension(s)? |
| 19:24.51 | fprior | ChannelZ, at the moment dialplan simply execute Read function |
| 19:26.06 | McBoingBo | ChannelZ, no I did not, a little intimidated messing with the outgoing contexts to be honest, thats why I was testing with my extension to see if I can get outgoing calls recorded |
| 19:26.10 | ChannelZ | fprior: then I suppose it depends on what arguments you gave Read and the definition of "no reaction". Pastebin your test extension and the console output of a test call with verbose turned on a bit |
| 19:26.27 | ChannelZ | McBoingBo: well then what you're getting is expected.. |
| 19:26.51 | McBoingBo | ChannelZ, once I have that down, I can attempt applying it to outgoing for all, I thought Set(__DYNAMIC_FEATURES=automon) would do the trick but didnt work |
| 19:27.47 | ChannelZ | no because that has no meaning |
| 19:29.18 | McBoingBo | ChannelZ, I was afraid you would say that, so how to proceed? my Set command needs to be on the actual _NXXXX.... exten? |
| 19:30.26 | navaismo | fprior, also enable the dtmf debug in the pastebin |
| 19:31.34 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 19:31.41 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:31.41 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:32.28 | rrittgarn | In 11, confbridge - i get an error when i try to set announce_join_leave=yes (says not a suitable category). Curious if the syntax is wrong in the example.conf file or if I'm missing something? |
| 19:33.57 | ChannelZ | McBoingBo: no you need the W or w on the Dial for that exten, just like you did with your 248 exten |
| 19:34.16 | McBoingBo | ChannelZ, thats it? |
| 19:34.24 | ChannelZ | yes |
| 19:34.41 | ChannelZ | Again, DYNAMIC_FEATURES are for user-created features |
| 19:35.19 | McBoingBo | ChannelZ, ok so no need, features.conf and Ww should suffice |
| 19:35.39 | ChannelZ | for automon yes |
| 19:37.05 | McBoingBo | ChannelZ, stumped by 2 W's lmao, thanks |
| 19:37.23 | ChannelZ | though again for your outgoing exten you might only want W |
| 19:37.44 | ChannelZ | unless you want the person you're calling to be able to start recording by hitting *1 as well (assuming they knew enough to try) |
| 19:38.02 | McBoingBo | yeah will only use one way, but the lowercase, is for other SIP useds right now just the "other end"? |
| 19:39.03 | McBoingBo | eek I thought it meant within my environment, like between 2 internal extens |
| 19:39.23 | ChannelZ | W = calling party (whoever initiated the Dial) w = called party (whomever got dialed) |
| 19:39.36 | McBoingBo | dangerous |
| 19:39.55 | ChannelZ | well it's only a recording, not like rm -rf or something :) |
| 19:41.49 | McBoingBo | woks like a charm, thanks again |
| 19:41.51 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 19:41.54 | jmetro | agi script to call rm-rf script |
| 19:41.54 | McBoingBo | works too :P |
| 19:41.57 | ChannelZ | but like I said the direction of the call matters so you need to set accordingly |
| 19:42.14 | McBoingBo | yeah took me so long to understand that |
| 19:42.34 | McBoingBo | so what if I am using exten 248, thats not what was called, so those parameters are not applied |
| 19:42.35 | ChannelZ | rrittgarn: where did you define it? |
| 19:42.47 | ChannelZ | an extension is not a phone |
| 19:43.39 | ChannelZ | an extension is just an arbitrary string of numbers (partial lie) that make Asterisk do something when they are dialed. |
| 19:43.59 | leifmadsen | Supporting documentation per ChannelZ comments: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id283201.html |
| 19:44.13 | rrittgarn | /etc/asterisk/confbridge.conf |
| 19:44.56 | ChannelZ | rrittgarn: I mean where.. that option is only applicable to a confbridge user, it sounds like maybe you are trying to set it under a bridge |
| 19:45.18 | rrittgarn | ah... that would be it then. |
| 19:45.28 | rrittgarn | Thank you sir. |
| 19:45.37 | ChannelZ | rrittgarn: IE under [something] type=user |
| 19:46.43 | fprior | ChannelZ , navaismo : http://pastebin.com/SADWVKZ6 |
| 19:48.25 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 19:49.11 | rrittgarn | yeah, I've just been using the default user, but the different bridges have different audio. So i figured the audio would be there. |
| 19:49.14 | ChannelZ | fprior: since you have no timeout, are you typing in 6 digits to test? (not going to dig through the RTP debug to count myself) |
| 19:51.34 | fprior | ChannelZ, yes, I typed 843575 and hangup myself after all |
| 19:52.54 | navaismo | fprior, can you remove the rtp debug and enable the dtmf debug, the phone from where you are dialing have the rfc2833 too? |
| 19:54.37 | fprior | navaismo, I've enabled dtmf from logger.conf in console section. Is enough ? No, phone_from have not rfc2833 |
| 19:55.09 | *** join/#asterisk Captain_Proton (~quassel@adsl-99-39-191-57.dsl.klmzmi.sbcglobal.net) |
| 19:56.20 | Captain_Proton | Quick ? for anyone how do I ring a ext with going to voicemail. I try to google but my search may have been wrong so I found no answer |
| 19:56.49 | ChannelZ | with going to? |
| 19:56.55 | Captain_Proton | sorry that should be without* |
| 19:57.15 | navaismo | yep thats enough, I cant see it in the pastebin, usaully i see somthing like: DTMF[22923][C-00000038] channel.c: DTMF end passthrough '5' on SIP/line1-00000017 |
| 19:57.22 | navaismo | fprior, ^ |
| 19:58.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:58.17 | Captain_Proton | Dial(SIP/${agent},30,rtT,novm) something like that? |
| 19:58.38 | navaismo | Captain_Proton, yep next priority hangup |
| 19:58.54 | ChannelZ | well generally speaking you just Dial with a timeout, and then in the next priority call Voicemail |
| 20:00.56 | ChannelZ | though I'm not sure what your 'novm' means (that's a URL parameter) |
| 20:01.20 | Captain_Proton | would the default time take over? so the phone would ring for 15 sec then move to the line? is my thinking correct? |
| 20:01.24 | *** join/#asterisk mardok45 (~mardok@74-140-29-110.dhcp.insightbb.com) |
| 20:01.43 | ChannelZ | Dial(SIP/whatever,30) would ring for 30 seconds before going on to the next priority |
| 20:01.54 | Captain_Proton | seen it online somewhere lol |
| 20:02.16 | ChannelZ | but otherwise yes |
| 20:02.23 | fprior | navaismo: pastebin.com/rpqnnprm without RTP. No dtmf traces from channel.c |
| 20:02.25 | Captain_Proton | i just want to show you what i was trying to do |
| 20:02.34 | Captain_Proton | thanks :) |
| 20:04.52 | ChannelZ | Generally you'd have something like exten => 123,1,Dial(SIP/Joe,15) followed by exten => 123,n,Voicemail(${EXTEN}@default,us) or somesuch |
| 20:05.29 | ChannelZ | 123, Joe and @default being as appropriate for you and possibly more. |
| 20:10.21 | ChannelZ | oh haha I missed your response about without voicemail |
| 20:10.34 | ChannelZ | that's what i get for doing too many things at once |
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| 20:13.04 | ChannelZ | I shal do something useful like reboot. |
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| 20:15.42 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 20:17.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:18.32 | mardok45 | I'm completely new to asterisk and am trying to set it up to use Google Voice, following this guid: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google I'm using Arch Linux and installed version 11 from the AUR. When I try to connect with SIP with empathy, it fails to connect and doesn't give me a reason why. There are no logs. How do I configure asterisk to start logging when I try to connect? |
| 20:18.39 | navaismo | fprior, seems like the server doesnt receive digits, now if you can enable the sip debug |
| 20:19.26 | mardok45 | Running nmap on the server shows that only port 22 is open |
| 20:19.34 | *** join/#asterisk fprior_ (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:20.06 | fprior_ | navaismo, I've lost session can you write again ? |
| 20:20.29 | ChannelZ | navaismo: the RTP debug did show the type 101 events |
| 20:20.53 | navaismo | ok iddint look at rtp debug... |
| 20:21.37 | ChannelZ | I eventually sifted through it to see |
| 20:28.55 | mardok45 | Now I can connect with a SIP client, but when I make a call, I get the message "Call from 'xxx' (ip:port) to extension '1xxxxxxx' rejected because extension not found in context 'local'." |
| 20:29.20 | ChannelZ | I dunno, is mixmonitor killing it? I notice the verbose doesn't match the example extension, it's doing other things. |
| 20:30.06 | ChannelZ | mardok45: that one says what it means, an extension doesn't exist in the local context that matches |
| 20:30.25 | ChannelZ | Did you literally just config the whole system based on that one wiki entry? |
| 20:31.29 | mardok45 | ChannelZ: Yes, what do I have to do next? |
| 20:32.07 | navaismo | fprior_, ChannelZ , using the pb dialplan it work but in the pb i see a mixmonitor as ChannelZ said |
| 20:34.15 | ChannelZ | I just tried with mixmonitor too and it still worked here |
| 20:35.45 | navaismo | yep --> http://pastebin.com/JWbxXzmv |
| 20:36.11 | navaismo | maybe the ITSP have issues with the dtmf |
| 20:37.43 | fprior_ | navaismo, ChannelZ, without MixMonitor is the same. Yes, I would say something to ITSP but what ? They say to have set dtmf mode to inband |
| 20:38.21 | [TK]D-Fender | markThat message clearly tells you what the call is looking for and where. Go make an extension there to match it. |
| 20:38.30 | fprior_ | I've tried with rfc2833compensate=yes, but no change |
| 20:38.59 | fprior_ | I would to be sure to have tested all from my side |
| 20:40.05 | navaismo | fprior_, the pastebin show the dtmfmode=rfc2833 if you change to inband that makes a difference? |
| 20:40.21 | [TK]D-Fender | mardok45, That message clearly tells you what the call is looking for and where. Go make an extension there to match it. |
| 20:40.33 | fprior_ | navaismo, nop |
| 20:47.16 | ChannelZ | mardok45: that example is very incomplete, and it's also only from the standpoint of GV.. so you calling in via SIP isn't handled at all |
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| 21:08.13 | ChannelZ | fprior_: sorry.. your ITSP is telling you they send DTMF inband? |
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| 21:16.22 | fprior_ | ChannelsZ, yes |
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| 21:24.15 | crazed1 | So we're still having to wait > 30 seconds some times for asterisk to respond to actions sent via ami (approx 1-3% of actions suffer this fate) so it happens frequently |
| 21:25.08 | leifmadsen | sounds like locking |
| 21:25.17 | leifmadsen | do you have DEBUG_THREADS or DONT_OPTIMIZE enabled? |
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| 21:26.10 | crazed1 | no |
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| 21:28.06 | leifmadsen | seems odd that AMI would ever wait 30 seconds to send a response, unless it is waiting for information from somewhere else |
| 21:28.26 | leifmadsen | think you'll need to show some examples and provide information as to what is actually taking the 30 seconds |
| 21:28.39 | crazed1 | sometimes we get the response event 20seconds later, with no additional data being sent |
| 21:29.07 | crazed1 | i could put that together, but here is what happens: |
| 21:29.38 | crazed1 | it tries to run the event, but if after 30 seconds no response, it, along with any other "failed" events, are resent immediately after closing and reopening the socket, and the events are processed immediately, identically |
| 21:30.19 | leifmadsen | system isn't blocking due to I/O? file descriptors maxed out? |
| 21:30.33 | leifmadsen | seems more like an issue outside asterisk to me, but I have nothing to base that on |
| 21:30.43 | Qwell | crazed1: Did you ever do a trace? |
| 21:31.51 | crazed1 | like strace? no i haven't really been able to learn that yet. iowait is very low, < 0.05%, plenty of cpu resources (asterisk with 200 users uses < 5%) |
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| 21:32.15 | WIMPy | pcap |
| 21:32.46 | Mon|A|rch | is there a predefined channel variable with the number of an incoming call? |
| 21:33.00 | leifmadsen | Mon|A|rch: use CALLERID() function |
| 21:33.10 | Mon|A|rch | ah |
| 21:33.11 | Mon|A|rch | cool |
| 21:33.13 | leifmadsen | points at asteriskdocs.org |
| 21:33.20 | leifmadsen | I think it's in the "Dialplan Basics" chapter |
| 21:33.28 | Mon|A|rch | i saw the callerid, it just didn't register, sorry |
| 21:33.47 | crazed1 | quick side-note, will asterisk automtaically bulid with pcap if libpcap-devel is installed, or does it need to be specified? |
| 21:34.17 | leifmadsen | o.O |
| 21:34.18 | WIMPy | use wireshark or whatever. |
| 21:34.22 | leifmadsen | that |
| 21:34.35 | crazed1 | no that was unrelated, it just got me thinking about it since i need to use pcap to mark packets also |
| 21:34.37 | leifmadsen | asterisk doesn't use libpcap |
| 21:36.16 | crazed1 | oh libcap, whoops |
| 21:36.24 | crazed1 | thats what i meant ;) |
| 21:40.20 | leifmadsen | jmetro: because no one has either modified app_queue to do what you're expecting, or you haven't configured it in such a way as to make things work as you expected. |
| 21:40.52 | leifmadsen | jmetro: if you care about time matching, then use the Local channel along with the functionality provided in order to perform time based or skills based routing |
| 21:41.06 | leifmadsen | jmetro: also, I'm replying here because you shouldn't msg people directly |
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| 21:47.19 | crazed1 | @leifmadsen should i compile with debug_threads and dont_optimize? |
| 21:47.23 | leifmadsen | no |
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| 22:29.50 | Greenlight | Evening all. I've a custom patch that I apply to my Asterisk builds, which creates an AMI event when we get SIP 183 Progress, as this is generally indicative that the lines started to ring, and there's no other way to tell this. I have the hook inside static struct ast_channel *wait_for_answer |
| 22:30.38 | Greenlight | Now, this only seems to fire when there is another channel "Dialling" the channel that's triggering the event. Where would be a better place to put this hook so it's always triggered ? |
| 22:31.39 | Greenlight | Basically I hook onto the event that creates the following message in CLI: ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in)); |
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| 22:52.03 | Greenlight | Maybe I can approach this from other direction; where abouts are the manager Newstate events generated ? |
| 22:53.05 | WIMPy | I think you should askt those questons in -dev. |
| 22:54.10 | Greenlight | Ahh okay I'll give that a try |
| 22:54.25 | Qwell | channel.c |
| 22:54.32 | Qwell | (grep is your friend) |
| 22:56.17 | Greenlight | Ok, thanks :) |
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| 22:56.37 | Greenlight | Guess I was grepping for the wrong things |
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| 23:22.53 | [TK]D-Fender | checkout time, BBIAB |
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