| 00:05.45 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net) |
| 00:24.50 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:27.22 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-135-101.dsl.stlsmo.sbcglobal.net) |
| 00:27.39 | LemensTS | When dialing a group of sip phones, is their a way to do BLF for only the one that answers? |
| 00:30.09 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:30.33 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:41.35 | *** join/#asterisk [Outcast] (~anonymous@pool-96-252-45-211.bstnma.fios.verizon.net) |
| 00:56.21 | [TK]D-Fender | LemensTS: Could you rephrase that.... doesn't make much sense as it was written... |
| 00:59.49 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 00:59.49 | *** mode/#asterisk [+o file] by ChanServ |
| 01:04.41 | LemensTS | TKD: On incoming calls from outside, I have it ring multiple phones 'Dial(SIP/800&SIP/802)' .....If I set 'hint,SIP/800&SIP/802' they both go ringing on the BLF which is good, but when one picks up they both go busy, instead of one going back to available. |
| 01:05.31 | LemensTS | I do the hint before the dial of course |
| 01:07.17 | dijib | p3nguin: sorry i had to eat dinner and cleanup |
| 01:07.37 | dijib | p3nguin: it says "starting asterisk" and then nothing... no [OK] |
| 01:07.53 | dijib | when i run ps i see it running but i cannot connect to asterisk -r |
| 01:08.01 | p3nguin | BLF is per extension, not per phone. Phones are not extensions, extensions are not phones. So it is working correctly. |
| 01:08.37 | p3nguin | dijib: What does "ps -C asterisk u" output? |
| 01:10.21 | LemensTS | p3nguin: yea I know it is working properly for how I have it written, how can I make it work where the one not answered goes back to available? |
| 01:16.11 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 01:19.57 | p3nguin | The extension is IN USE. BLF reports it as such. |
| 01:21.27 | p3nguin | dijib: Eating again? |
| 01:23.39 | LemensTS | p3nguin. yes I know it reports it as because of the hint i send it. But only 1 of those phones answers, the other one quits ringing. Yes I know that i show it is supposed to work and it works by extension and not phone. I want to know how to write it where when the one picks up, the other one goes available on BLF. I figured their is a fancy way to do this in the dialplan using dev_state or something... |
| 01:23.42 | dijib | no |
| 01:23.55 | dijib | the user when i run it with service is asterisk |
| 01:23.58 | dijib | 500 |
| 01:24.23 | LemensTS | p3nguin: or maybe I should be calling the phones using a different method idk |
| 01:24.59 | dijib | hold on ive got to make a call |
| 01:26.27 | [TK]D-Fender | LemensTSTKD: On incoming calls from outside, I have it ring multiple phones 'Dial(SIP/800&SIP/802)' .....If I set 'hint,SIP/800&SIP/802' they both go ringing on the BLF which is good, but when one picks up they both go busy, instead of one going back to available. <- that's the hint. it refers to multiple devices |
| 01:27.02 | [TK]D-Fender | LemensTS: It isn't a device that's stuck... its that "only wanting to see if all are busy" etc expectation. That isn't going to happen unfortunately... |
| 01:27.19 | [TK]D-Fender | LemensTS: You'd have to do some other trickery for that sort of thing. The messy kind |
| 01:28.40 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 01:35.22 | LemensTS | TKD: ok, do most people not mess with BLF on incoming calls to SIP phones? |
| 01:35.42 | LemensTS | In ring groups I mean |
| 01:37.12 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:38.11 | p3nguin | dijib: <p3nguin> dijib: What does "ps -C asterisk u" output? |
| 01:41.27 | dijib | root 31634 5.9 5.2 1484064 26052 pts/0 Sl 21:47 0:00 /usr/sbin/asterisk -f -vvvg -c |
| 01:41.37 | dijib | its starting as root? |
| 01:41.55 | dijib | <PROTECTED> |
| 01:48.02 | p3nguin | Also set runuser and rungroup to asterisk in asterisk.conf |
| 01:52.41 | dijib | root 32049 11.6 5.2 1483564 26040 pts/0 Sl 21:58 0:00 /usr/sbin/asterisk -f -vvvg -c |
| 01:52.47 | dijib | after the asterisk.conf addition |
| 01:53.07 | dijib | should that be in options? |
| 01:53.11 | dijib | context? |
| 01:55.20 | p3nguin | Context? We're not talking about dial plan. You don't even have asterisk running right yet. |
| 01:56.03 | p3nguin | In asterisk.conf, under the options heading, did you set runuser = asterisk and rungroup = asterisk? |
| 02:01.21 | [TK]D-Fender | LemensTS: I don't know who uses BLF to monitor a group of phones expecting to see only an "all busy / some ringing state. |
| 02:01.29 | [TK]D-Fender | LemensTS: * simply doesn't work that way. |
| 02:01.55 | [TK]D-Fender | LemensTS: You could fake your own way through it by using a a custom DEVICE_STATE and a whack of dialplan. As I said... messy... |
| 02:04.06 | *** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za) |
| 02:07.10 | dijib | asterisk 32523 1.4 5.3 1484060 26148 pts/2 Sl 22:12 0:00 /usr/sbin/asterisk -f -vvvg -c |
| 02:07.14 | dijib | ok still cant connect |
| 02:07.18 | dijib | but its running |
| 02:08.10 | dijib | configs are definitly not running |
| 02:14.33 | p3nguin | At least it is running as asterisk, now. Now to check permissions. |
| 02:14.48 | p3nguin | What happens when you run asterisk -r ? |
| 02:14.53 | p3nguin | dijib: ^ |
| 02:14.54 | dijib | oui |
| 02:15.15 | dijib | hold on a teck... |
| 02:15.20 | dijib | this is not making sense |
| 02:15.27 | dijib | its running. let me restart again |
| 02:16.25 | dijib | [Mar 21 22:22:32] Running as group 'asterisk' |
| 02:16.25 | dijib | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 02:16.48 | dijib | also the digium vs of res_fax_digium.so was causing it to crash |
| 02:17.00 | dijib | the res_fax_digium.so is working |
| 02:17.17 | dijib | but ya no right now i still cannot connect |
| 02:17.54 | p3nguin | namei -m /var/run/asterisk/ |
| 02:17.58 | p3nguin | What does that say? |
| 02:18.11 | p3nguin | It should return five lines. |
| 02:18.25 | dijib | all drwxr-xr-r |
| 02:18.47 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:19.28 | p3nguin | I really dislike the namei on CentOS. It fails to reveal the ownership of the path. |
| 02:19.40 | p3nguin | ls -dl /var/run/asterisk |
| 02:20.20 | dijib | owned by asterisk |
| 02:20.48 | p3nguin | You started asterisk using the init script or service asterisk start? |
| 02:21.24 | dijib | service. |
| 02:21.27 | p3nguin | You are root when running asterisk -r? |
| 02:21.32 | dijib | yes |
| 02:21.57 | p3nguin | I think I might see a problem. |
| 02:22.03 | dijib | what? |
| 02:22.10 | p3nguin | chmod 775 /var/run/asterisk |
| 02:22.54 | p3nguin | Although, I can't see how that could make it not connect. |
| 02:22.57 | dijib | still no go |
| 02:23.02 | p3nguin | Yeah, I figured. |
| 02:23.09 | dijib | its also not running the dialplan |
| 02:23.28 | dijib | asterisk 564 1.6 5.2 1483568 26088 pts/2 Sl 22:28 0:00 /usr/sbin/asterisk -f -vvvg -c |
| 02:23.36 | dijib | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 02:23.45 | dijib | and the pid does exist but the ctl does not |
| 02:27.34 | p3nguin | I have one more idea. I'm comparing your ps output to a CentOS box of mine. |
| 02:28.04 | p3nguin | Mine explicitly runs asterisk using the -U and -G options. That is going to be a result of the init script. |
| 02:28.55 | p3nguin | You're sure you set AST_USER=asterisk and AST_GROUP=asterisk in /etc/sysconfig/asterisk? |
| 02:29.07 | dijib | yes |
| 02:29.21 | dijib | whats the /etc/asterisk/asterisk.conf variables |
| 02:29.46 | dijib | runuser rungroup |
| 02:30.51 | p3nguin | I want you to check the init script to ensure the values are being read. Pastebin the entire output of: grep "sysconfig\|USER\|GROUP" /etc/init.d/asterisk |
| 02:32.24 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:32.58 | dijib | http://pastebin.com/y7uVyxSr |
| 02:33.38 | dijib | no such file or directory |
| 02:33.53 | dijib | ? |
| 02:33.54 | p3nguin | Well, that is a result of your doing it wrong. I see the info I wanted to see. |
| 02:34.36 | p3nguin | Restart it using the init script, not service. /etc/init.d/asterisk restart |
| 02:35.06 | dijib | is that whats run on boot or the service? |
| 02:35.15 | p3nguin | The init script runs. |
| 02:35.28 | Tim_Toady | btw is selinux disabled? |
| 02:35.33 | p3nguin | But I really thought service executed the init script. |
| 02:35.55 | Tim_Toady | service runs the init script actually |
| 02:36.05 | p3nguin | I thought it did. |
| 02:36.13 | *** join/#asterisk SeRi (~wtf@c-98-200-177-50.hsd1.tx.comcast.net) |
| 02:36.24 | Tim_Toady | dijib: is selinux disabled? |
| 02:36.38 | dijib | http://pastebin.com/yRxUmB1S |
| 02:36.41 | p3nguin | You'll probably have to tell him how to check it. I doubt he knows anything about it. |
| 02:36.47 | dijib | selinux ? whats this? |
| 02:36.53 | dijib | i still cant connect |
| 02:37.00 | dijib | different output this time |
| 02:37.03 | dijib | on that grep |
| 02:37.23 | Tim_Toady | dijib: try running 'sestatus' and see if selinux is enabled |
| 02:37.26 | Tim_Toady | if yes disbale it |
| 02:37.26 | dijib | selinux disabled where Tim_Toady |
| 02:37.35 | dijib | eneables. |
| 02:37.37 | dijib | eneabled |
| 02:37.59 | p3nguin | Uh... oops? :/ |
| 02:38.01 | Tim_Toady | disbale it, it causes problems with asterrisk |
| 02:38.52 | dijib | how do you disable it? |
| 02:39.00 | dijib | what else will that impact? |
| 02:39.10 | Tim_Toady | edit /etc/selinux/config |
| 02:39.11 | dijib | its a security policy? |
| 02:39.21 | p3nguin | It's a system, not a policy. |
| 02:39.31 | Tim_Toady | and change SELINUX=enforcing to SELINUX=disabled |
| 02:40.03 | dijib | how do i restart that service? |
| 02:40.07 | dijib | or whatever it is? |
| 02:40.16 | dijib | ive changed the parameter |
| 02:40.20 | Tim_Toady | reboot if possible |
| 02:40.56 | dijib | can do but im going to have to reconnect to irssi |
| 02:41.55 | dijib | back in 5 |
| 02:42.04 | dijib | while watching doomsday preppers |
| 02:42.57 | Tim_Toady | u can also run setenforce 0 rto avoid rebooting |
| 02:44.05 | p3nguin | It's probably too late. |
| 02:44.19 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 02:44.44 | Tim_Toady | a reboot is always better |
| 02:45.01 | Tim_Toady | he might never make it back here :P |
| 02:45.09 | p3nguin | In the case of selinux, I'd feel better knowing it was booted with it disabled. |
| 02:45.21 | p3nguin | For almost everything else, rebooting just wastes my time. |
| 02:47.55 | Tim_Toady | root... |
| 02:48.26 | p3nguin | Yep. |
| 02:48.53 | Tim_Toady | not even selinux can save him :P |
| 02:49.24 | *** join/#asterisk v4x (~v4x@d209-89-85-139.abhsia.telus.net) |
| 02:49.53 | p3nguin | Maybe you can tell him when he gets back. |
| 02:50.04 | *** join/#asterisk dijib (~root@bas10-kitchener06-1279682470.dsl.bell.ca) |
| 02:51.38 | *** join/#asterisk nosaj (~jbarinas@186.85.225.82) |
| 02:52.21 | dijib | asterisk 1610 0.2 5.3 1485240 26472 ? Sl 22:53 0:00 /usr/sbin/asterisk -f -vvvg -c |
| 02:52.28 | dijib | and asterisk -r is working |
| 02:52.30 | dijib | after reboot |
| 02:52.52 | p3nguin | So it was selinux blocking asterisk from writing the ctl file. |
| 02:53.13 | p3nguin | And asterisk is running as asterisk, as it should be, instead of root. |
| 02:53.26 | p3nguin | irssi, on the other hand, is still running as root. |
| 02:53.30 | dijib | (y) |
| 02:53.49 | p3nguin | So you're almost there. |
| 02:53.50 | dijib | so selinux is a kernel protecton module? |
| 02:54.01 | dijib | yeh now i just need res_fax.so working |
| 02:54.21 | p3nguin | en.wikipedia.org/wiki/SELinux |
| 02:54.41 | dijib | then ConfBridge, Meetme, voicemail, vm=email, mixmonitor and IVR are working |
| 02:55.58 | p3nguin | I have to stop trusting that people disable selinux first thing. |
| 02:56.46 | dijib | well i just built this thing |
| 02:56.52 | dijib | im not that savvy i guess |
| 02:56.56 | dijib | sorry yall |
| 02:57.36 | dijib | so this crashes http://downloads.digium.com/pub/telephony/fax/res_fax/asterisk-1.6.2.0/x86-64/res_fax-1.6.2.0_1.3.0-x86_64.tar.gz |
| 02:57.39 | dijib | what can i do |
| 02:58.00 | dijib | 2.6.32-220.7.1.el6.x86_64 |
| 02:58.38 | Tim_Toady | i would first suggest you to upgrade to asterisk 1.8 |
| 02:58.52 | Tim_Toady | 1.6 is unsopported |
| 02:58.55 | dijib | im running 1.8.7 |
| 02:59.09 | dijib | Connected to Asterisk 1.8.7.0 currently |
| 02:59.14 | Tim_Toady | thes this module is failing because its the 1.6.2 version |
| 02:59.19 | Tim_Toady | get the 1.8 one |
| 02:59.26 | dijib | and if there isnt one? |
| 02:59.30 | Tim_Toady | there is |
| 02:59.38 | dijib | where? |
| 02:59.42 | p3nguin | hahahaha |
| 02:59.49 | Tim_Toady | in the same place u got this |
| 02:59.59 | dijib | negative. |
| 03:00.16 | p3nguin | Running 1.8.7.0, using a module CLEARLY made for 1.6.2.0... wondering why it crashes. Brilliant! |
| 03:00.36 | Tim_Toady | http://downloads.digium.com/pub/telephony/fax/res_fax_digium/asterisk-1.8.4/ |
| 03:00.38 | Tim_Toady | posotive |
| 03:00.49 | Tim_Toady | positive even |
| 03:00.53 | p3nguin | http://downloads.digium.com/pub/telephony/fax/README |
| 03:01.01 | p3nguin | Should have read the README. |
| 03:01.21 | dijib | ive got the 1.8 version of res_fax_digium.so |
| 03:01.29 | dijib | can i use the stock res_fax.so? |
| 03:01.33 | Tim_Toady | thats all you need to run it |
| 03:01.34 | Tim_Toady | yes |
| 03:01.38 | dijib | or do u need ti digium version? |
| 03:01.42 | dijib | oh ok |
| 03:01.43 | p3nguin | Considering it has all the necessary instructions and links, the README is pretty important. |
| 03:01.56 | dijib | then thats working.. but im not able to recieve faxes |
| 03:02.15 | dijib | i must have missed it |
| 03:02.24 | Tim_Toady | welcome to the wonderful world of faxing :P |
| 03:02.32 | dijib | i did the 1,3,4 |
| 03:02.39 | p3nguin | You have to have res_fax_digium.so for it to work. |
| 03:02.48 | dijib | i have it |
| 03:02.50 | p3nguin | http://www.digium.com/en/docs/FAX/faa-download.php |
| 03:02.55 | dijib | and its running im fairely certain |
| 03:03.21 | p3nguin | fax show capabilities |
| 03:03.27 | dijib | they dont have a 64bit version for 1.8? |
| 03:03.34 | p3nguin | Yes they do. |
| 03:03.45 | dijib | Type : DIGIUM |
| 03:03.45 | dijib | Description : Digium FAX Driver |
| 03:03.46 | dijib | Capabilities : SEND RECEIVE T.38 G.711 MULTI-DOC |
| 03:03.51 | p3nguin | If you'd use the module selector, you'd know that. |
| 03:04.24 | p3nguin | Did you register your key? |
| 03:04.24 | dijib | there is a 64bit version on the ftp. |
| 03:04.27 | dijib | yes |
| 03:04.45 | dijib | fax show stats shows licences. |
| 03:05.05 | dijib | in g.711 t.38 |
| 03:05.21 | p3nguin | But does fax show licenses show your license? |
| 03:05.42 | dijib | yes it shows |
| 03:06.08 | p3nguin | I guess it works, then. |
| 03:06.12 | p3nguin | Next? |
| 03:06.58 | dijib | tsting faxing now. |
| 03:07.01 | dijib | testing |
| 03:08.03 | dijib | next is raid on the 80gb's, then addition of 1.5TB drive |
| 03:10.20 | dijib | ...and then there was silence in #asterisk |
| 03:13.01 | p3nguin | Way to kill an IRC channel. |
| 03:14.19 | dijib | haha |
| 03:14.25 | dijib | dont worry im still rocking it |
| 03:14.50 | dijib | faxes are failing |
| 03:15.32 | dijib | -- Channel 'SIP/voipms-00000001' FAX session '1' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT' |
| 03:16.50 | dijib | also the script says its writing a file to /var/spool/asterisk/fax/filename but no file is ever created |
| 03:18.04 | dijib | so what can i do with this sangoma card also.... |
| 03:18.14 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:19.36 | dijib | Channel 'SIP/voipms-00000002' receiving FAX '/var/spool/asterisk/fax/20120321-8009806858-swissarms-1332386711.2.tif' |
| 03:21.33 | dijib | no file ever really created |
| 03:21.36 | dijib | permissions are good |
| 03:21.44 | dijib | ls -Als |
| 03:24.40 | p3nguin | Who is sending the faxes? |
| 03:24.47 | Tim_Toady | thats because the fax never arrived, T1 timeout means a failure during fax negotiation |
| 03:25.39 | dijib | im using thos automated internet services |
| 03:25.57 | dijib | enter sender/reciver info and msg |
| 03:26.06 | dijib | faxzero, got freefax, etc |
| 03:27.02 | Tim_Toady | t1 timeout means the faxes failed to identify each other |
| 03:27.24 | Tim_Toady | you try to get the fax via pstn? sip? |
| 03:29.22 | dijib | dont have access |
| 03:30.11 | dijib | whats a good web based file manager? |
| 03:30.39 | dijib | i dont like extxplorer |
| 03:32.39 | p3nguin | Don't have access to what? |
| 03:33.14 | p3nguin | And what would you do with a web-based file manager, anyway? |
| 03:33.31 | dijib | i dont have access to a pstn fax machine |
| 03:34.02 | dijib | i have some drives ive mounted to /mnt directories that i need to access and mv files around |
| 03:34.18 | Tim_Toady | install mc and do it in cli |
| 03:35.00 | dijib | hmmm thats more like what im looking for |
| 03:35.06 | dijib | i would rather do it from a browser though |
| 03:36.15 | p3nguin | You won't use a web browser to do it. It isn't the "web." |
| 03:36.45 | p3nguin | But you can use ssh or sftp based apps, such as WinSCP or Filezilla. |
| 03:39.22 | dijib | does that keep the cpu load at the server? |
| 03:39.36 | dijib | i dont want client processing/bandwidth |
| 03:40.04 | dijib | also sambs & windows7 you guys attempted to map in windows? |
| 03:40.26 | dijib | i can connect but it wont enumerate any files. gives permission error when going into the share |
| 03:40.59 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 03:40.59 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:41.29 | p3nguin | Maybe you can better define what it is you are trying to accomplish. |
| 03:45.09 | *** join/#asterisk mahaD (~chatzilla@123.238.235.241) |
| 03:45.36 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 03:45.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:45.40 | dijib | mapping a network drive to a samba share on the server |
| 03:52.02 | p3nguin | Okay, that's pretty basic and very simple, but has nothing to do with your web based idea. |
| 03:52.32 | p3nguin | Install samba. Configure a share. Map a drive on a client. |
| 03:53.42 | dijib | ok now the share is a mount of a hard drive |
| 03:54.40 | dijib | i have [sharename] |
| 03:54.42 | dijib | path |
| 03:54.45 | dijib | valid users |
| 03:54.50 | dijib | read ony =no |
| 03:54.56 | dijib | create mast 0777 |
| 03:55.05 | dijib | directory mask 0777 |
| 03:55.27 | dijib | can connect from windows box but can not enumerate anything in the share. |
| 03:55.45 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 03:55.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:56.04 | *** join/#asterisk flyingbull (~Adium@cpe-065-190-158-078.nc.res.rr.com) |
| 03:56.16 | flyingbull | Hi everyone:) |
| 03:56.58 | flyingbull | Question, I'm trying to figure out why I can call out fine, and the person who gets the call can hear everything, but I don't hear the person I called. |
| 03:58.07 | flyingbull | I'm using Asterisk 1.8 and the weird thing is, I can receive calls just fine, and transfer them without a problem. |
| 03:58.22 | dijib | flyingbull thats a NAT issue with forwarding 5060 appropriety |
| 03:58.45 | flyingbull | Yeah but the server isn't behind a nat. |
| 03:58.47 | dijib | also some routers firmware is not so SIP friendly |
| 03:59.02 | dijib | its got a public ip without a firewall? |
| 03:59.09 | dijib | whats the topology? |
| 03:59.36 | flyingbull | it is on a server, CentOS, I have turned off Ip tables at the moment to see if that was the problem, doesn't appear to be. |
| 03:59.51 | flyingbull | it has a static ip. |
| 04:00.01 | dijib | your directly connected to the internet ? |
| 04:00.38 | p3nguin | Not hearing one side of the call has nothing to do with forwarding of port 5060. |
| 04:00.48 | flyingbull | I am, via a router, but I've had this problems with other systems, and I setup the DMZ on the router to allow any connections directly to me. So I'm wide open. |
| 04:01.16 | p3nguin | ~dmz |
| 04:01.16 | infobot | [~dmz] De-Militarized Zone, or usually a separate physical or logical network that has limited access to your internal systems and is accessible in limited ways from untrusted networks such as the Internet. Putting Asterisk in the DMZ is not an acceptable alternative to properly forwarding the appropriate ports, so don't do it. Plastic router appliances generally do not implement DMZ well. |
| 04:01.41 | dijib | word up infobot |
| 04:01.57 | flyingbull | LOL |
| 04:01.59 | p3nguin | One-way audio is almost always a problem with NAT configuration. |
| 04:02.49 | p3nguin | Since your asterisk system is not behind a NAT, perhaps your phones that you are testing with are. |
| 04:03.00 | flyingbull | Some of them but not all. |
| 04:03.09 | flyingbull | but if it is a nat, maybe it is my sip configuration — one moment. |
| 04:03.14 | p3nguin | Then you'll have to configure asterisk for nat support. |
| 04:03.50 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 04:04.21 | flyingbull | Well the template that I use, is set with nat=yes. |
| 04:04.56 | dijib | whats your router and firmware? |
| 04:05.02 | dijib | is it a plastic router? |
| 04:06.07 | flyingbull | no, actually its a dedicated server, I seem to remember them telling it was a cisco — They pretty much set it up as wide open, I had to put on the firewall and everything on there. |
| 04:06.10 | p3nguin | I'm wondering about the statement where you've used the phrase "via a router." What exactly do you mean by that? |
| 04:07.29 | flyingbull | I am currently on a home network, with a router to the internet. A linksys router, and I set it up to forward everything to this computer directly. |
| 04:07.46 | p3nguin | That's where asterisk resides? |
| 04:08.09 | flyingbull | No, the Asterisk server is a dedicated server I rent from a company in Arizona actually. |
| 04:08.35 | p3nguin | What does this home computer have to do with it? Is that where you have a soft phone for testing? |
| 04:09.37 | dijib | hey are you running ddwrt on that linksys router flyingbull ? |
| 04:09.48 | flyingbull | No. |
| 04:09.56 | dijib | stock firmware? |
| 04:10.07 | flyingbull | p3nguin — I was answering a previous question as to where I was, in realtionship to the asterisk system. |
| 04:10.35 | p3nguin | You said you're forwarding ports to your computer, though. What does that have to do with it? |
| 04:10.48 | dijib | he said he was using dmz |
| 04:10.57 | p3nguin | That's mistake #1. |
| 04:11.10 | p3nguin | But I want to know why he's forwarding ports to his home computer. |
| 04:11.12 | dijib | any way to build a software raid on the fly? |
| 04:11.22 | p3nguin | Sure. |
| 04:11.30 | dijib | the port forwarding is just to the server. |
| 04:11.35 | dijib | sure how? |
| 04:11.43 | dijib | i dont fully believe that statement |
| 04:11.50 | flyingbull | Actually I said I was using the DMZ setting, it was that so I was completely wide open to the asterisk system — this was so I was outside of the nat issue. |
| 04:11.52 | p3nguin | It's not something that can be answered in a single statement. |
| 04:12.03 | dijib | lpstat |
| 04:12.10 | p3nguin | I've done it, so I know it can be done. |
| 04:12.30 | flyingbull | brb |
| 04:12.37 | p3nguin | You're still not giving me the whole picture. |
| 04:13.02 | p3nguin | There's a reason you felt like you needed to forward ports or assign DMZ to a system. I want to know the reason. |
| 04:13.15 | p3nguin | Once you tell me the reason, I can tell you why it's wrong. |
| 04:13.47 | dijib | lol |
| 04:13.52 | flyingbull | p3nguin — the reason I did the DMZ on my home system, was so that I wasn't getting cluttred with the NAT issue, if that was it. |
| 04:13.54 | flyingbull | brb |
| 04:14.05 | dijib | how do i umount / & /boot to build a raid while the system is running |
| 04:14.05 | dijib | ? |
| 04:14.06 | p3nguin | Okay, your system is STILL behind NAT. |
| 04:14.22 | p3nguin | Like it or not, you are behind the NAT. |
| 04:14.55 | dijib | flyingbull: its easy. give * box a static private Ip and forward 5060 & 10000-10099 to it |
| 04:14.56 | p3nguin | And I still don't see what your reason for wanting to remove NAT from your PC is. That's the part you continue to leave out. |
| 04:15.14 | p3nguin | 10000-10099? Where did you come up with this random set of numbers? |
| 04:15.47 | flyingbull | Because I was trying to determine if my * system wasn't working with the phone issue, because of NAT. |
| 04:15.49 | dijib | from the 10000-20000 minus 9901 |
| 04:16.08 | p3nguin | Is your phone on the computer that you've erroneously put in the DMZ? |
| 04:16.12 | p3nguin | soft phone |
| 04:17.52 | flyingbull | Yes. |
| 04:18.01 | p3nguin | Okay, that was the piece you kept leaving out. |
| 04:18.24 | flyingbull | But only in the last hour have I put it on the DMZ - to see if I could get the voice oneway issue to go away. |
| 04:18.39 | p3nguin | Remove any DMZ settings; don't forward ports on the phone side of the equation. |
| 04:19.06 | p3nguin | Ensure that the soft phone is not configured to do its own nat traversal or nat mapping. |
| 04:20.15 | flyingbull | ok. brb |
| 04:20.48 | p3nguin | With asterisk on a public IP address, you don't need to configure externaddr/externhost, but you do need to make sure you have nat=yes set so that asterisk can figure out the phone's public IP address rather than its private one. |
| 04:24.36 | flyingbull | p3nguin: I have nat=yes under the sip template, Should I put that in the general section of the sip.conf file? |
| 04:25.20 | flyingbull | When I say sip template, I have a standard template, that I set all my settings, then refer to it in my other sip sections. |
| 04:25.22 | p3nguin | I don't know what your template applies to, but putting it in the general section should be okay in any case. |
| 04:25.53 | dijib | his asterisk will be behind the nat on a prvate ip |
| 04:26.04 | dijib | and what usefullness can i do with this sangoma card |
| 04:26.08 | p3nguin | His asterisk is on a public IP address. |
| 04:26.46 | p3nguin | But he has phones behind NAT. |
| 04:27.48 | flyingbull | yeah, ok, so I put in the general settings nat=yes. did a sip reload, and still the same problem. |
| 04:28.22 | p3nguin | Since you said you were applying it via template, I am not surprised at no change. |
| 04:28.57 | p3nguin | I'm interested in seeing a sip debug of a call with one-way audio. |
| 04:30.54 | flyingbull | Verbose or something a bit more detailed? |
| 04:30.59 | p3nguin | Have you set directmedia=no? |
| 04:31.22 | flyingbull | no. |
| 04:31.31 | p3nguin | Set that and see if it helps. |
| 04:31.44 | flyingbull | Under general? |
| 04:32.02 | p3nguin | Yes. And make sure you do not override it in any peer involved in this test. |
| 04:32.03 | flyingbull | scp asterisk@ |
| 04:32.10 | p3nguin | ? |
| 04:32.12 | flyingbull | sorry wrong window. |
| 04:32.14 | flyingbull | lol |
| 04:32.31 | p3nguin | How could that be a good command for any window? |
| 04:32.47 | flyingbull | because I hit enter by accident instead of backspace. |
| 04:32.50 | p3nguin | ssh asterisk ... vim /etc/asterisk/sip.conf |
| 04:33.09 | flyingbull | scp does the trick pretty quickly for me actually. |
| 04:33.15 | p3nguin | gross |
| 04:33.56 | flyingbull | When I used to ssh in, for some reason the window would overwrite and garble the text — only on the mac, so I started just doing it locally, then uploading it. |
| 04:34.00 | p3nguin | I guess you and dijib ought to get along real well. |
| 04:35.07 | flyingbull | well shit that did the trick. |
| 04:35.52 | p3nguin | Ah, good. |
| 04:35.59 | dijib | garbage day tomorrow |
| 04:36.08 | flyingbull | Thank you very much p3nguin. I've been like mystified by this issue for a couple of days now. |
| 04:36.30 | dijib | yeah flyingbull just give me ssh access and ill set you straight |
| 04:36.44 | p3nguin | spews |
| 04:36.55 | dijib | p3nguin: my own has been pleguing me for at least a week now |
| 04:37.01 | dijib | selinux... heff |
| 04:37.22 | p3nguin | He who IRCs as root shall never touch the system of another. |
| 04:37.24 | flyingbull | You know it says in the Asterisk book, like the 4 thing they do is turn off Selinux. |
| 04:38.10 | flyingbull | 4th or 5th i know it on that list of things you do when you are setting up the system. |
| 04:38.46 | flyingbull | Anyway, thank you very much for you help, got to go hang out with my son, he has been bugging me to play skyward sword for the last 20 minutes LOL |
| 04:39.48 | p3nguin | A satisfied customer. |
| 04:40.00 | p3nguin | That's one in a row! |
| 04:46.15 | dijib | im not satisfied. |
| 04:46.30 | dijib | dude im so satisfied i have a smoke in my mouth |
| 04:46.56 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wzgpadxqhoigfout) |
| 04:50.36 | dijib | are you allowed to barter p3nguin > |
| 04:50.36 | *** part/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net) |
| 04:51.13 | p3nguin | I... guess. I'm not sure who has the authority to disallow me from doing what I want to do. |
| 05:07.48 | dijib | oh ok |
| 05:08.08 | dijib | then what do u got for this fxo? |
| 05:08.31 | dijib | didnt Seri send you some random atas? |
| 05:08.36 | dijib | or those were lost in the mail? |
| 05:09.52 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 05:10.39 | citywok | So i can NOT figure out why my phone doesn't work from home, but did when at work. If i bind the same username/extension to an aastra phone it works, but my UT670 does not. I just get 401 Unauthorized. thoughts? http://pastebin.com/DaF2WY0h |
| 05:11.14 | citywok | i set core debug to 10, verbose to 10, and the debug log provides no help, neither does the verbose log. |
| 05:14.44 | dijib | no clue |
| 05:16.03 | p3nguin | I have no use, other than for education, for the card. And seri never sent me the stuff he said he would send me. |
| 05:17.11 | dijib | if only i had a couple of the fxs modues then it would bo of sme use |
| 05:17.20 | citywok | bah panasonic needs to learn how to make a phone that doesn't suck |
| 05:18.02 | citywok | how do i get Asterisk to tell me why it is rejecting a phone from registering? is that possible? it just kicks out 401 unauth |
| 05:19.12 | p3nguin | If only the configuration had something to do with phones registering. |
| 05:21.28 | citywok | lol p3nguin, the phone worked at work on this extension, but doesn't work at home. another phone works at home on the same extension. |
| 05:21.45 | citywok | one can deduce the problem lies with the phone |
| 05:21.54 | p3nguin | Phones don't care about extensions. |
| 05:21.57 | citywok | and seeing how digium certified the phone for asterisk it _should_ work |
| 05:22.06 | citywok | s/extension/peer/ |
| 05:22.24 | p3nguin | I'd imagine it is related to the configuration. |
| 05:22.38 | citywok | safe bet |
| 05:23.01 | p3nguin | So far, all you've confirmed is that things work on your local network. |
| 05:23.33 | p3nguin | Check domain names, IP addresses, registration strings, auth users, etc. |
| 05:23.35 | citywok | so far i've confirmed the phone works at the office, and a phone made by aastra bound to the same peer with the same config works from my house, but the panasonic doesn't. asterisk says 401unauth to the pana |
| 05:23.57 | citywok | yea i've retyped everything 37 times, it's not a simple ip typo or auth user. especially considering the same config worked at work :p |
| 05:24.14 | citywok | to be fair at work the DNS resolves to the internal address, and from home it resolves to the external |
| 05:24.17 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 05:24.23 | p3nguin | But the phones are DIFFERENT. |
| 05:24.31 | citywok | really? |
| 05:24.37 | bkruse | ~karma bkruse |
| 05:24.37 | infobot | bkruse has neutral karma |
| 05:24.50 | p3nguin | If you could dump config from one working phone and apply it to another, then you could say how you're using the same configs. |
| 05:25.20 | citywok | oh, my apologies. |
| 05:25.38 | citywok | i'm using the exact same settings in the exact same "relative" fields. aka sip userid, password, and server |
| 05:26.12 | p3nguin | You've proven that the peer entry is good for a local phone. Did you prove that the peer entry is good for a "working" phone when installed remotely? |
| 05:26.18 | citywok | dude |
| 05:26.20 | citywok | i said that 3 times |
| 05:26.28 | citywok | an aastra phone with the same peer works from my house just fine |
| 05:26.35 | citywok | the panasonic phone with the same peer does not work |
| 05:26.52 | citywok | the panasonic phone with the same exact config as i used at home (which did not work), works at work |
| 05:27.04 | dijib | gnite boys gotta go to bed, court tomorrow |
| 05:27.09 | dijib | wish me luck |
| 05:27.18 | citywok | dijib: gl |
| 05:27.18 | p3nguin | You're treating me like I'm familiar with your deployment and I know what you have and where you have it. |
| 05:27.23 | citywok | [22:24:19] <citywok> so far i've confirmed the phone works at the office, and a phone made by aastra bound to the same peer with the same config works from my house, but the panasonic doesn't. asterisk says 401unauth to the pana |
| 05:27.41 | citywok | [22:11:23] <citywok> So i can NOT figure out why my phone doesn't work from home, but did when at work. If i bind the same username/extension to an aastra phone it works, but my UT670 does not. I just get 401 Unauthorized. thoughts? http://pastebin.com/DaF2WY0h |
| 05:27.42 | dijib | thanks again p3nguin |
| 05:27.58 | citywok | twice before i said the same thing. 3 combinations, 2 work. |
| 05:28.07 | dijib | i'll try and stop in and add some insight for anons |
| 05:28.07 | p3nguin | Don't get thrown in the pokey for contempt! |
| 05:28.23 | dijib | yeh right, i didnt lie at all... the other guy has :D |
| 05:28.37 | dijib | and ive submitted proof |
| 05:28.42 | citywok | i just walked in and said yes, i did it, and plead out to reckless driving :p |
| 05:29.02 | dijib | not good, i had that once... flipped the old bmw |
| 05:29.12 | citywok | i didn't flip my car but blew a .15 |
| 05:29.29 | dijib | is there no decent webmin asterisk module? |
| 05:29.29 | citywok | the scary part is i remember most of that night, i can't imagine what i would have blown all the nights i drove home and have no recollection of it |
| 05:29.36 | dijib | al i can find is thirdlane |
| 05:29.48 | p3nguin | dijib: No, because a typical asterisk admin does not require such crap. |
| 05:29.50 | citywok | dijib: i haven't used webmin since redhat 8 years ago |
| 05:30.25 | dijib | well i like having all that server info at my fingertip. i cant remember every command like you can |
| 05:30.31 | dijib | peabrained |
| 05:31.34 | p3nguin | There's always FreePBX for that type. |
| 05:33.37 | dijib | yeh no... i would rather build the server and be hands o |
| 05:33.38 | dijib | n |
| 05:33.52 | citywok | i can see it being useful |
| 05:33.59 | dijib | i would just rather have an at a glance web service |
| 05:34.06 | citywok | for our lower level techs i had to make a couple interfaces so they could do basic functions |
| 05:34.20 | citywok | without having to fully understand how asterisk works inside out |
| 05:36.39 | dijib | see im not a company im just a home user so my needs are less |
| 05:37.26 | dijib | k anyways gnite gents |
| 05:39.37 | citywok | Does anybody have any experience with the Panasonic SIP Phones, or specifically the UT670? |
| 05:47.35 | *** join/#asterisk MrOli (~oli@ip70-187-135-51.oc.oc.cox.net) |
| 05:50.57 | MrOli | hello all |
| 05:51.31 | citywok | hello |
| 05:52.07 | MrOli | anyone kows a way to convert a string to all-lowercase ? something that would look like Set(MyVar=${LOWER(${ThisVar}}) ? |
| 05:52.45 | citywok | ${TOLOWER()} |
| 05:53.17 | citywok | p3nguin: i enabled transport=tcp,udp and set the phone to TCP and it works. lol. fail panasonic |
| 05:54.50 | MrOli | thanks citywok |
| 06:05.46 | citywok | MrOli: np |
| 06:38.15 | *** join/#asterisk jkroon (~jkroon@dsl-244-32-68.telkomadsl.co.za) |
| 06:40.55 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 06:56.13 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:09.53 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:21.01 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:21.04 | schmidts | good morning |
| 07:22.43 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 07:36.51 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:41.11 | j0b | aight. listen up |
| 07:41.23 | j0b | fuck im getting tired of this |
| 07:41.39 | ChannelZ | yawns |
| 07:41.50 | j0b | i guess i have to draw a model of what we are trying to do |
| 07:42.11 | j0b | this is getting on my nervs, and i would appriciate some input here |
| 07:42.34 | j0b | i will do this in gimp haha |
| 07:42.38 | j0b | free draw |
| 07:43.14 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:44.15 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:49.43 | ChannelZ | I have no idea what you're even going on about |
| 07:52.24 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:59.08 | schmidts | ChannelZ thats what j0b said ;) |
| 08:00.19 | ChannelZ | Clearly I've missed a huge chunk of conversation |
| 08:08.58 | wdoekes2 | we all have |
| 08:09.35 | ChannelZ | probably for the better |
| 08:11.33 | MrOli | amen to that |
| 08:12.06 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 08:14.01 | schmidts | :D |
| 08:17.19 | *** join/#asterisk radic (~radic@dslb-178-002-209-011.pools.arcor-ip.net) |
| 08:19.13 | *** join/#asterisk radic (~radic@dslb-178-002-209-011.pools.arcor-ip.net) |
| 08:20.28 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 08:22.08 | j0b | http://www.qfpost.com/file/d?g=YtqkhYLZ9 |
| 08:22.19 | j0b | can someone give me some tip here |
| 08:22.28 | j0b | its a pdf that describes the whole flow |
| 08:22.31 | j0b | sort of;) |
| 08:22.47 | j0b | i have messed with this like 20 hours |
| 08:22.53 | j0b | i shit you not:( |
| 08:22.57 | j0b | http://www.qfpost.com/file/d?g=YtqkhYLZ9 |
| 08:23.20 | j0b | and its not in gimp, its a dia(uml) |
| 08:23.26 | j0b | converted to pdf |
| 08:23.27 | j0b | http://www.qfpost.com/file/d?g=YtqkhYLZ9 |
| 08:34.35 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a07d3.pool.mediaWays.net) |
| 08:34.54 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:35.27 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:42.15 | *** join/#asterisk af_ (~getsmart@78.134.18.236) |
| 08:51.19 | *** join/#asterisk timahvo1 (~rogue@197.176.11.234) |
| 08:57.29 | *** join/#asterisk jofry (~procube@port-212-202-156-85.static.qsc.de) |
| 09:11.13 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:35.26 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:40.07 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:43.24 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:47.12 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:03.16 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 10:03.23 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:27.17 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 11:09.49 | jacc0 | @j0b: not may people are going to download the pdf, especialy while you are not explaining your problem, you sound like a spam-bot that is trying to infect people with some uploaded pdf |
| 11:11.33 | jacc0 | if you want people to have a look at it you did better convert it to jpg |
| 11:12.02 | jacc0 | en explain what your problem is here |
| 11:12.10 | jacc0 | *and explain |
| 11:14.13 | *** join/#asterisk Szaszka (~szaszka@catv-178-48-182-66.catv.broadband.hu) |
| 11:18.48 | *** join/#asterisk makmak78 (~makmak78@83.145.38.138) |
| 11:19.34 | makmak78 | hello! im using asterisk 1.4.36. im confused regarding sip cause to reason mapping, anybody could shed some light on this |
| 11:19.38 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:20.24 | makmak78 | im getting reason: 1 when the number is invalid. this reason code should be noanswer if you check voip.org |
| 11:21.02 | makmak78 | does anyone know where to find these mappings in the source code |
| 11:25.08 | wdoekes2 | makmak78: you'll probably want to check ${HANGUPCAUSE} .. that maps to the values in causes.h |
| 11:27.42 | makmak78 | alright |
| 11:30.31 | makmak78 | but i would like to view in the sourcecode what cause is mapped to what reason |
| 11:31.45 | wdoekes2 | makmak78: check chan_sip.c for the sip2cause and cause2sip functions |
| 11:31.55 | makmak78 | Okay |
| 11:34.25 | makmak78 | i cant find where it says reason: 8 is mapped to cause 27 for example |
| 11:34.56 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 11:34.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:37.56 | *** join/#asterisk timahvo1 (~rogue@197.176.11.234) |
| 11:39.04 | wdoekes2 | makmak78: if you mean DIALSTATUS, it's set in app_dial.c |
| 11:41.37 | *** join/#asterisk krotos (~d3v1l@host249-34-dynamic.2-87-r.retail.telecomitalia.it) |
| 11:41.41 | krotos | hi all :) |
| 11:42.30 | krotos | i' have a simple question. if i sett on a peer with type=friend , the option "call-limit=10", the limit on both incoming and outgoing calls is 10? |
| 11:42.54 | krotos | because i don't understand what mean "limitonpeer" options |
| 11:43.04 | krotos | i don't say if i have to use this or not |
| 11:47.07 | makmak78 | wdoekes2: i will check that , tanks |
| 11:51.08 | *** join/#asterisk eicto (~morgoth@144-71.dsl.aichyna.com) |
| 11:55.58 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net) |
| 12:01.39 | *** join/#asterisk hhkahya (~hulusi@77.79.99.28) |
| 12:05.07 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:07.00 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
| 12:07.05 | StaRetji | hi folks |
| 12:07.33 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:07.50 | StaRetji | I just had some accident with server and I luckily I had backup of conf file. However, conf files are from asterisk 1.4 and new server version is 1.8 |
| 12:08.22 | StaRetji | I see sip trunks are okay, but users can't connect to sip server |
| 12:08.32 | *** join/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 12:08.44 | StaRetji | [Mar 22 13:06:23] WARNING[1736]: chan_sip.c:25515 set_insecure_flags: Unknown insecure mode 'very' on line 169 |
| 12:08.59 | catphish | does anyone know a good way to dispose of the bodies of customers who won't stop using the phrase 'hunt group'? |
| 12:09.03 | [TK]D-Fender | StaRetji, "insecure=port,invite" |
| 12:09.36 | StaRetji | thx [TK]D-Fender |
| 12:09.40 | StaRetji | I will edit now |
| 12:14.32 | StaRetji | that fixed the error, however non of clients in user_sip.conf wont connecte except 1 |
| 12:14.47 | StaRetji | I looked and conf is exactly the same as for other users |
| 12:15.06 | StaRetji | so, I don't know what it could be, why would 1 user connect and other don't |
| 12:16.23 | StaRetji | has maybe #include "user_sip.conf" changed in new version? |
| 12:17.44 | [TK]D-Fender | nope |
| 12:18.11 | [TK]D-Fender | StaRetji, otherwise NONE of them would have loaded. |
| 12:18.12 | StaRetji | hm, I renamed fils to user_sip.confX |
| 12:18.17 | StaRetji | and core reload |
| 12:18.24 | StaRetji | asterisk doesn't complain |
| 12:18.28 | StaRetji | that file is missing |
| 12:18.38 | [TK]D-Fender | StaRetji, You should be looking at SIP DEBUG at *CLI to see what is coming in and what * is responding.... |
| 12:18.52 | *** part/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 12:18.55 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 12:19.19 | StaRetji | I need to fix this asap, I'm willing to pay for support |
| 12:19.51 | StaRetji | I'm to excited can't concentrate lol |
| 12:20.08 | schmidts | staretji sorry i missed whats your problem ;) |
| 12:20.22 | schmidts | but pay for something allways sounds good ;) |
| 12:20.52 | StaRetji | hehe |
| 12:21.11 | StaRetji | I pasted conf files from version 1.4 to version 1.8 |
| 12:21.26 | StaRetji | and I excepcted it should work |
| 12:21.45 | StaRetji | but it wont, it only connects to sip peers (sip providers) |
| 12:22.06 | StaRetji | and I have auto_sip.conf where I keep sip settings for clients |
| 12:22.22 | StaRetji | however, it seems that those settings are ignored... |
| 12:23.13 | schmidts | do you see any warnings errors when you do a sip reload? |
| 12:23.51 | StaRetji | Section 'default' lacks type |
| 12:27.38 | schmidts | section default? |
| 12:28.00 | schmidts | this warning normally occurs only for sip peers when you miss the type entry there but you should not have a sip peer in default |
| 12:33.53 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:35.14 | StaRetji | Unable to register extension '8003', priority 2 in 'default', already in use |
| 12:35.18 | StaRetji | I see this |
| 12:35.19 | StaRetji | hm |
| 12:39.09 | StaRetji | <PROTECTED> |
| 12:39.40 | StaRetji | I mean, this works on 1.4 |
| 12:40.49 | StaRetji | my God |
| 12:40.53 | StaRetji | is comment still ; |
| 12:40.54 | StaRetji | ? |
| 12:41.09 | StaRetji | I have double lines, of course, but they are commented with ; |
| 12:46.50 | *** join/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 12:47.58 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 12:49.40 | rampage73 | anyone have good advice on getting faxes to work with asterisk? I am currently getting about 50% of faxes if they are a single page and 20% if they are multi page, I have lowered the baud rate of fax to 9600 that seemed to help but not enough |
| 12:50.12 | catphish | rampage73: are you using a hangup extension to process the fax? |
| 12:51.01 | rampage73 | i can but not always sometimes the fax is also the telephone with voicemail |
| 12:51.53 | catphish | i had similar numbers of fax failures but it seemed to be the result of trying to process and email the fax file which was getting aborted because the remote hund up |
| 12:52.03 | catphish | if you do the processing in a h extension, it doesn't get aborted |
| 12:52.16 | catphish | it's easy enough to set a variable to the code doesn't run on voice calls |
| 12:53.19 | [TK]D-Fender | no... |
| 12:53.29 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:53.33 | [TK]D-Fender | If you're in "h" .. the call is ALREADY DEAD |
| 12:53.36 | rampage73 | catphish, easy for gurus :) I know enough to be dangerous! |
| 12:53.47 | [TK]D-Fender | h = hangup. You don't receive ANYTHING in there |
| 12:54.00 | catphish | [TK]D-Fender: that's the point, you don't do the processing until the call is ended |
| 12:54.32 | catphish | you receive the fax in the call, then process it after hangup |
| 12:54.38 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 12:54.58 | [TK]D-Fender | catphish, He's not even getting the whole fax due to other issues. post processing is irrelevant, as is "h" |
| 12:55.15 | pigpen | I love "h" |
| 12:55.27 | catphish | [TK]D-Fender: if you're sure, i'm just commenting that i had a similar issue and that was the problem |
| 12:55.30 | *** join/#asterisk flyingbull (~Adium@cpe-065-190-158-078.nc.res.rr.com) |
| 12:55.39 | rampage73 | sorry did not mean to cause a war ! |
| 12:55.44 | pigpen | catphish, he is sure. |
| 12:55.47 | catphish | i was receiving faxes fine, but they "appeared" to get lost because i was processing them during the call |
| 12:56.05 | pigpen | I do all kinds of cool stuff in the "h", but it is AFTER the call is hung up. |
| 12:56.11 | catphish | indeed |
| 12:56.15 | [TK]D-Fender | Once you are out of whatever fax app you're using you either have it or you don't. Nothing after that app matters |
| 12:56.20 | catphish | emailing faxes is an ideal example |
| 12:56.39 | catphish | [TK]D-Fender: is does if you're measuring success by whether you get an email or not :) |
| 12:56.44 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:56.50 | catphish | which was my problem |
| 12:57.06 | [TK]D-Fender | catphish, Do you see him saying it's simply not e-mailing? |
| 12:57.13 | catphish | of course rampage73 we'd be more helpful if we had an error from your log |
| 12:57.32 | [TK]D-Fender | It's also be helpful if we knew anything at all about the call itself... |
| 12:57.53 | rampage73 | catphish, I will try and get what you need will take time though as I do not have a fax machine available to test with at the moment |
| 12:57.56 | [TK]D-Fender | And what fax app. And what version of *. |
| 12:58.00 | catphish | rampage73: send a log snippet and a dialplan :) |
| 12:58.13 | rampage73 | [TK]D-Fender, thanks I will get some info and be back |
| 12:58.24 | rampage73 | catphish, ok |
| 12:58.25 | rampage73 | thank you both |
| 12:58.34 | rampage73 | be back soon |
| 12:58.50 | *** part/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 12:59.37 | [TK]D-Fender | And... we don't even get the slightest bit of real details.... niiiiices |
| 13:00.28 | catphish | lol |
| 13:00.37 | catphish | he'll be back (maybe) :) |
| 13:02.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:02.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:06.39 | *** join/#asterisk e-Zee (ezee@besuch.mal.bestfails.com) |
| 13:07.15 | catphish | is there anything in the AMI to manage voicemail? |
| 13:07.36 | catphish | or should one just move the files around? |
| 13:08.16 | [TK]D-Fender | catphish, nothing in AMI. |
| 13:08.26 | catphish | where is the metadata stored? |
| 13:09.11 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
| 13:09.39 | *** join/#asterisk serafie1 (~erin@nat/digium/x-ggnsyudiijjtmivj) |
| 13:09.50 | StaRetji | please, anyone willing to help me fast, I will pay, thx |
| 13:10.03 | StaRetji | I don't know to fix it myself |
| 13:11.32 | StaRetji | <PROTECTED> |
| 13:11.59 | StaRetji | I didn't change extensions.conf file, just copy pasted from 1.4 to 1.8 |
| 13:12.08 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 13:12.10 | StaRetji | in 1.4 works. in 1.8 don't :/ |
| 13:13.36 | chuckf | StaRetji: did you read the changes between the different versions to see what may have changed for settings you have in extentions.conf? |
| 13:14.11 | StaRetji | no, I didn't |
| 13:14.26 | *** join/#asterisk sohoindra (4dd15c07@gateway/web/freenode/ip.77.209.92.7) |
| 13:14.46 | chuckf | StaRetji: the quickest fix would be for you to read those changes that might affect your conf file |
| 13:14.50 | sohoindra | Hello to everyone |
| 13:15.16 | StaRetji | thx chuckf, I will try, I have 30 more minutes |
| 13:15.25 | StaRetji | then I can forget it :/ |
| 13:15.57 | sohoindra | I have 2 systems with the same hardware and software configuration connected to an E1 multiplexer |
| 13:16.35 | chuckf | StaRetji: what happens in 30 minutes? |
| 13:16.38 | sohoindra | one systems works fine but the other is geting LMFA/OK in all the spans of the E1 card |
| 13:17.20 | sohoindra | I would like to know the meaning of this alert in order to solve the problem |
| 13:17.25 | sohoindra | thanks in advance |
| 13:17.38 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 13:18.39 | StaRetji | in 30 minutes, old server gets unplugged |
| 13:18.47 | StaRetji | and taken away |
| 13:18.49 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 13:19.03 | StaRetji | and I'm left with this installation that doesn't work = I get fired |
| 13:19.05 | StaRetji | lol |
| 13:19.29 | StaRetji | and the guy who administers is not around, so, I'm stuck |
| 13:19.29 | catphish | why not just use 1.4 until you've tested a new config |
| 13:19.40 | StaRetji | how, I have ubuntu server |
| 13:19.46 | StaRetji | installed it via apt-get |
| 13:19.50 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pauptktwigzchgsz) |
| 13:19.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:19.52 | StaRetji | so, installed 1.8 |
| 13:19.56 | catphish | eek |
| 13:19.58 | chuckf | don't let them take the old server |
| 13:19.59 | StaRetji | yep |
| 13:20.11 | catphish | remove it, install one from source, easy on ubuntu |
| 13:20.12 | StaRetji | well, it's not in my power |
| 13:20.29 | StaRetji | will it remove all sound files? |
| 13:20.33 | StaRetji | there are so many |
| 13:20.39 | chuckf | are you really saying this company is willing to trash their entire phone system if you can't get this working in 30 minutes or less? |
| 13:20.40 | StaRetji | I lost 2 hours checking them |
| 13:20.42 | catphish | yes perhaps |
| 13:20.43 | StaRetji | custom ivr :/ |
| 13:20.58 | StaRetji | no, the thing is, it was planned |
| 13:21.01 | catphish | why the rush |
| 13:21.08 | catphish | why didnt you test this config days ago? |
| 13:21.13 | StaRetji | but the guy that was suppose to migrate is not online |
| 13:21.23 | StaRetji | yep, I trusted another person |
| 13:21.27 | StaRetji | stupid, I know |
| 13:21.50 | catphish | the issue is likely with your sip peers, but i couldn't tell you why |
| 13:21.54 | StaRetji | anyway, I have asterisk connected to providers, but now of clients can't connect |
| 13:22.09 | StaRetji | none of* |
| 13:22.39 | catphish | i wonder how messy it will be to move to asterisk 10 |
| 13:22.47 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 13:23.05 | catphish | then again, long term support sounds tasty too |
| 13:23.06 | chuckf | catphish: can it be any worse than it is now? |
| 13:23.21 | catphish | chuckf: i mean the process of moving over :) |
| 13:23.34 | StaRetji | I set core set verbose 15 |
| 13:23.46 | chuckf | catphish: oh, I thought you meant for StaRetji's situation |
| 13:23.48 | StaRetji | but still there is nothing, like nothing comes to the server |
| 13:23.55 | FinboySlick | This isn't entirely #asterisk related... But does anyone here have experience with large-ish deployment of SPA-2102 ? |
| 13:24.29 | catphish | StaRetji: if you see nothing at verbose 15 then likely the phones are misconfigured |
| 13:24.44 | catphish | but to be totally sure you can enable sip debug for the IP of your phone |
| 13:25.18 | chuckf | StaRetji: are the phones pointing to the correct server? |
| 13:25.22 | StaRetji | yes |
| 13:25.27 | StaRetji | ip just changed |
| 13:25.29 | catphish | anyway i'm a little busy, shouldn't be distracting myself here |
| 13:25.38 | StaRetji | ok, thx |
| 13:25.49 | *** join/#asterisk ollii (~risker@a81-14-178-55.net-htp.de) |
| 13:25.54 | FinboySlick | I'm having strange issues with 1/10 of them completely ignoring tftp provisioning (straight out of the box). |
| 13:28.15 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:30.39 | catphish | do they not do http? |
| 13:32.30 | leifmadsen | anyone happen to have a lab Polycom device that they could do a test for me? I just want to know if, in the <mac_address>.cfg file, if you are specifying files to load, that if you have a trailing comma with no filename after, if it fails to boot |
| 13:35.03 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:35.03 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:35.29 | *** join/#asterisk wesphillips (~wphill04@212.39.180.45) |
| 13:36.26 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:37.54 | catphish | where is voicemail metadata stored? |
| 13:39.36 | leifmadsen | /var/spool/asterisk/voicemail/ |
| 13:40.53 | catphish | oh yeah, its alongside the files |
| 13:40.54 | catphish | thanks! |
| 13:41.11 | FinboySlick | catphish: I'm pretty sure they do http as well. My beef was mostly the inconsistency... Some working some not straight out of the shrinkwrap. |
| 13:41.45 | catphish | i've never thought of tftp as terribly reliable, but it does seem odd |
| 13:41.51 | catphish | do the failed one just never work? |
| 13:41.54 | catphish | after a restart etc |
| 13:42.16 | catphish | and are they misconfigured? or just faulty? |
| 13:42.20 | FinboySlick | Well, if we provision them by hand through the web interface, they work fine. |
| 13:42.39 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
| 13:42.42 | catphish | well of course, but that doesn't help with the question |
| 13:42.59 | catphish | you really want to work out if they're a) unreliable b) faulty or c) misconfigured |
| 13:43.10 | FinboySlick | They just never ask the tftp for the config files. (tcpdump shows that they request the tftp option and are given the right reply through dhcp) |
| 13:43.46 | catphish | and the devices that don't what about after a reboot? |
| 13:43.58 | catphish | are they permenently failing? |
| 13:44.15 | FinboySlick | They never do, reboot or no reboot. Another new device from the same box will work as expected. |
| 13:44.47 | catphish | have you checked their config |
| 13:45.00 | catphish | to see if the 10% are configured differently |
| 13:45.05 | catphish | or have different firmware? |
| 13:45.09 | FinboySlick | Well, they all came with the factory config, they're from the same crate. |
| 13:45.18 | catphish | have you actually checked that? |
| 13:45.38 | catphish | it's not impossible they're from different batches, or even refurbed? |
| 13:45.57 | catphish | clearly their software is somehow different |
| 13:46.22 | FinboySlick | It's possible. I was mostly trying to establish if it's a relatively common occurrence or if I had to lay out all the diagnosis groundwork on my own. |
| 13:46.48 | catphish | afraid i've never used them myself, i'm just trying to throw ideas about :) |
| 13:47.08 | FinboySlick | catphish: I'm actually grateful for that. |
| 13:47.20 | catphish | then again refurb isn't that likely, they're incredibly cheap aren't they? |
| 13:47.46 | catphish | actually no, they're not the ones i was thinking of |
| 13:48.11 | catphish | i was thinking of the pap2t |
| 13:48.45 | catphish | anyway, i'd compare the firmware version, and config of a good and a failing one |
| 13:48.51 | catphish | and see if there are any differences |
| 13:48.56 | catphish | if not, just complain :) |
| 13:56.21 | *** join/#asterisk emate (~marcin@81.219.183.142) |
| 13:56.33 | emate | Hi guys |
| 13:58.45 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:59.36 | *** join/#asterisk bitwize (~chatzilla@h87-96-213-2.dynamic.se.alltele.net) |
| 14:01.49 | emate | I have little problem with calls recording. When i try to play recorded call, it is a kind of speeded up record - it doesn't matter if i use gsm/wav/wav49 format. I tried to play it with sox's play and mplayer - no success. |
| 14:04.41 | *** join/#asterisk Kroms (~Kroms@72.37.252.50) |
| 14:12.30 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 14:12.51 | *** join/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 14:14.22 | hurdman | hi, i try to have a sort of "synchronization" between to call on the same asterisk server, is there something like a thread join or a module for ? |
| 14:20.28 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
| 14:29.40 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 14:31.40 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:34.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:34.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:36.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:45.14 | *** join/#asterisk sereal (~serealz@unaffiliated/sereal) |
| 14:46.36 | sereal | Do some asterisk applications (like system() ) get ran in their own thread? I'm wondering because I am doing some fax -> email stuff and I think the script is being ran while the fax is incoming. |
| 14:47.23 | sereal | So if this is the case would doing a hangup() followed by a system() make sure that system() isn't called before the fax is complete? (I am assuming hangup() will always wait for the fax to come threw) |
| 14:49.09 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
| 14:50.39 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:54.12 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:56.51 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.74.253) |
| 14:57.40 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.103) |
| 14:58.12 | kaldemar | sereal: System starts a new process and it waits for the command to exit unless it is started in background. |
| 14:58.22 | kl4m | sereal: I don't think system(...) will ever execute if you hangup before |
| 14:58.48 | kaldemar | sereal: if you put a hangup in your dialplan, nothing after it in the same extension gets executed. |
| 14:58.56 | sereal | humm okay. but if say a fax is coming in, does asterisk try and run the next line in the dial plan while that is happening |
| 14:58.58 | chuckf | StaRetji: still employed? |
| 14:59.08 | *** join/#asterisk xertres (~xertres@h-198-78.a176.priv.bahnhof.se) |
| 14:59.26 | kaldemar | sereal: and hangup really hangs up the call, it will not wait for anything. |
| 14:59.26 | StaRetji | yep, server shut down |
| 14:59.40 | StaRetji | but new not working lol still |
| 15:00.37 | chuckf | so the company has no phones? |
| 15:00.56 | [TK]D-Fender | <sereal> humm okay. but if say a fax is coming in, does asterisk try and run the next line in the dial plan while that is happening <- no |
| 15:01.37 | sereal | [TK]D-Fender, Okay thanks. It is clearly something wrong with my script. It's a bitch to debug since I gotta ask people to send me faxes, wait then debug, then ask to send another :P |
| 15:05.33 | j0b | http://www.qfpost.com/file/d?g=YtqkhYLZ9 |
| 15:06.06 | j0b | can someone (who know what they are talking about) tell me whats going on here |
| 15:06.11 | j0b | again: http://www.qfpost.com/file/d?g=YtqkhYLZ9 |
| 15:06.18 | j0b | or give a hint |
| 15:06.25 | j0b | or several hints |
| 15:06.46 | j0b | would be appriciated |
| 15:12.05 | *** join/#asterisk Kroms (~Kroms@72.37.252.50) |
| 15:15.14 | kaldemar | j0b: pastebin your real configs configs and CLI output of a call with verbosity and sip debug enabled. |
| 15:17.05 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a07d3.pool.mediaWays.net) |
| 15:19.59 | [TK]D-Fender | ~pb |
| 15:20.00 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:20.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 15:21.27 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:22.00 | j0b | kaldemar: well, its all there |
| 15:22.23 | j0b | used pastebin |
| 15:22.33 | *** join/#asterisk chasing`Sol (~cS@197.134.243.141) |
| 15:22.38 | kaldemar | and put the link to an image in a pdf... |
| 15:22.45 | j0b | right |
| 15:22.52 | [TK]D-Fender | j0b, Considering there are dozens of sites offering that service it'd be nice for you to give us the LINK to your precise post |
| 15:22.54 | j0b | image? |
| 15:23.15 | j0b | yes, the links are in the image |
| 15:23.21 | j0b | or pdf |
| 15:23.24 | j0b | that is |
| 15:24.02 | j0b | [TK]D-Fender: but ok |
| 15:24.10 | j0b | i will give it directly to you |
| 15:24.12 | j0b | waut |
| 15:24.14 | j0b | wait |
| 15:27.11 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:27.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:28.38 | *** join/#asterisk timahvo1 (~rogue@41.81.28.188) |
| 15:31.21 | p3nguin | *sigh* |
| 15:35.33 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-060.mobile.uci.edu) |
| 15:40.52 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a07d3.pool.mediaWays.net) |
| 15:49.43 | LemensTS | How do you make Asterisk use a different sip port in Asterisk 10? I set udpbindaddr=0.0.0.0:65085 in general context, and port=65085 in users context, reload sip and reboot phone but it still shows 5060 when i do sip show peers? |
| 15:50.59 | LemensTS | btw these are polycom phones pulling cfg files off the server, i have reg.1.server.1.port="65085" set in the cfg files as well |
| 15:52.39 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 15:54.23 | leifmadsen | LemensTS: you probably need to unload then load the sip module for that kind of change to take affect |
| 15:54.32 | leifmadsen | effect.. |
| 15:54.52 | LemensTS | leifmadsen: does restarting asterisk do that? |
| 15:55.05 | leifmadsen | restart, yes |
| 15:55.06 | leifmadsen | not reload |
| 15:55.16 | LemensTS | ok, I have done that and it didn't help |
| 15:59.38 | LemensTS | netstat -nap shows 'udp 0.0.0.0:65085 asterisk' .... but sip show peers shows 5060 as their port |
| 16:01.20 | LemensTS | Nevermind, I see that means 5060 on the phone device.... |
| 16:01.32 | p3nguin | The bind port is asterisk's listening port. port is the client port. Decide which one you need to change. If you change the client port, you have to also change the sip port of the device's client IN ADDITION TO the server port. |
| 16:01.36 | kaldemar | LemensTS: sip show peers shows the port that the peers use in their end. |
| 16:02.30 | LemensTS | p3nguin/kaldemar: thanks, i understand now |
| 16:02.39 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:03.09 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 16:10.09 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:10.39 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 16:14.52 | *** join/#asterisk flyingbull (~Adium@cpe-065-190-158-078.nc.res.rr.com) |
| 16:21.39 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:22.09 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 16:24.13 | *** join/#asterisk philippel_mac (~p_lindhei@c-24-19-228-236.hsd1.wa.comcast.net) |
| 16:26.49 | *** join/#asterisk uluatu (~denis@187.59.34.205) |
| 16:28.57 | *** join/#asterisk navaismo (~navaismo@fixed-203-203-179.iusacell.net) |
| 16:29.06 | *** join/#asterisk moy (~moy@216.172.42.74) |
| 16:35.27 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 16:47.09 | *** join/#asterisk timahvo1 (~rogue@41.80.238.21) |
| 16:54.00 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 16:55.01 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 17:33.45 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:39.38 | uluatu | ping |
| 17:42.33 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 17:47.53 | Nugget | pong |
| 17:52.14 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 17:53.09 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 17:55.18 | uluatu | When an atendant try to transfer, atxfer, a call and he can't reach the destination number, the call returns and when he try to do t again, asterisk tells him he doesen't know the dialed number. |
| 17:56.35 | uluatu | This occur because the call now has a new context due the fact that the fist try to do the transfer hit the h extension on the previous context. And this h extenios sent him to another context that doesn't have access to the desired transfer number . |
| 17:56.52 | uluatu | Ive upgraded from 1.4.20.1 to the 1.4.42 |
| 17:57.06 | uluatu | 1.4.20.1 didn't have this problem. |
| 17:57.16 | uluatu | Is there anything new that I should know about? |
| 17:58.38 | autofsckk | a good sip client for android? |
| 17:58.40 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:59.11 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 17:59.46 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:24.28 | p3nguin | uluatu: It's a mistake in your configuration, which you failed to provide so we can show you where the error is. It has nothing to do with the upgrade in version. |
| 18:29.05 | *** join/#asterisk adeel|work (~adeel@unassigned-220.80.183.216.net.blink.ca) |
| 18:30.38 | adeel|work | if a variable is set in a parent context via Set(FOO="bar"), can i test that variable in a sub-context? does it need to be __FOO? |
| 18:30.51 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 18:31.01 | *** join/#asterisk serafie1 (~erin@75.76.38.159) |
| 18:31.48 | kaldemar | adeel|work: variables are tied to channels, not contexts. |
| 18:31.52 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:31.56 | p3nguin | adeel|work: Underscores are for channel inheritance. A single underscore allows the variable to be inherited to one new channel spawning from the parent channel. Two underscores allow the variable to be inherited through multiple levels of new channels. |
| 18:32.26 | adeel|work | ah....hmmm...then my logic in this gotoif must be wrong |
| 18:32.28 | adeel|work | thanks |
| 18:33.03 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:33.41 | p3nguin | If you think it is right but it doesn't work, pastebin it and someone will probably tell you why they think it isn't working. |
| 18:34.41 | adeel|work | i think i just need to group my conditions better.... |
| 18:35.08 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 18:36.31 | uluatu | p3nguin: ok, I take a deep dive into conf files. Today is the first day using this version. BTW it didn't happened with the old version. |
| 18:36.32 | *** join/#asterisk mattpattie (~matt@static-94-72-244-73.karoo.kcom.com) |
| 18:36.49 | p3nguin | Something changed in your configuration. |
| 18:37.15 | mattpattie | Hi guys very quick question. would you use asterisk for LLU in uk? |
| 18:39.10 | p3nguin | Sure. Why not? |
| 18:40.54 | p3nguin | If regulations do not disallow it, and if it helps in profiting your business, I'd go for it. |
| 18:42.03 | p3nguin | Here in the US, regulation insists that the owner if infrastructure MUST make the infrastructure available to competitors. |
| 18:42.12 | mattpattie | its something ive been asked to look into. after recently binning a VERY Expensive voip-pbx in favor for asterisk i was woundering if it was capable to support a lot of callers for a small city :) |
| 18:42.16 | p3nguin | s/if/of the/ |
| 18:42.49 | adeel|work | is the syntax for this conditional correct? http://pastebin.com/AzcXUNEA furthermore, what conditions would cause that gotoif to execute the true branch? |
| 18:43.06 | p3nguin | Asterisk alone will probably not be suitable for a local telco. Asterisk, in addition to other tools... maybe. |
| 18:43.36 | p3nguin | You're probably wanting more of a soft switch, such as FreeSWITCH. |
| 18:44.39 | mattpattie | not really looked into free switch? |
| 18:45.40 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 18:45.57 | p3nguin | adeel|work: If that will even work at all, I think your first three conditions will all have to be true, OR your last condition could be true, and it will go to label 40. |
| 18:47.07 | mattpattie | i'll have a nosy and a play Thankyou P3nguin |
| 18:47.24 | p3nguin | mattpattie: With asterisk and the right tools, you could make it go. Look into OpenSIPS or Kamailio as a proxy to put in front of your Asterisk back-end. |
| 18:47.25 | adeel|work | p3nguin, ok, well, oddly enough, it still somehow matches, even though LOCALTOOUTSIDE=1 |
| 18:47.54 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 18:48.09 | p3nguin | To be honest, I have never tried to combine both & and | in my If conditions. |
| 18:48.52 | p3nguin | Is THISISIVR currently not foo, and is OUTSIDETOINTERNAL currently not 1? |
| 18:48.59 | adeel|work | mattpattie, if you do some research on kamailio/opensips, there are some slides talking about an enterprise setup in germany i believe |
| 18:49.24 | adeel|work | p3nguin, yup, those 2 aren't set (since its an outbound call, which why LOCALTOOUTSIDE=1) |
| 18:49.39 | mattpattie | Cheers guys. i'll let you know how i get on :) |
| 18:49.47 | adeel|work | mattpattie, they claim a fairly substantial call volume |
| 18:49.48 | p3nguin | So LOCALTOLOCALOUT is null. |
| 18:50.06 | adeel|work | p3nguin, yes |
| 18:50.28 | mattpattie | time to pick up the misses ttfn |
| 18:50.34 | p3nguin | SER is most certainly capable of handling massive call volume. I'd imagine OpenSER/OpenSIPS is the same. |
| 18:52.18 | p3nguin | I don't know why the expression is coming back with the wrong value. Maybe it isn't capable of such complex expression. That's like a double-compound expression. |
| 18:53.01 | adeel|work | hmmm....let me try simplifying it and seeing what happens |
| 18:53.21 | p3nguin | Maybe split it into two If statements and see if you get the result you want. |
| 18:53.23 | adeel|work | p3nguin, should i wrap the different tests in $[]? |
| 18:53.34 | p3nguin | Probably. |
| 18:53.39 | p3nguin | I don't know if the ( ) work right. |
| 18:53.54 | p3nguin | Like I mentioned, I have never tried to do what you are doing there. |
| 18:54.53 | p3nguin | like so http://pastebin.com/H2J1339H |
| 18:55.00 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-nvyrfqkbtujdjbab) |
| 18:55.36 | p3nguin | If it is capable of performing the test within the test, that might just do it. |
| 18:55.49 | p3nguin | It could also cause horrible failure. |
| 18:56.02 | adeel|work | heh |
| 18:56.35 | adeel|work | what exactly does the first part of that test mean, "foo${THISISIVR}" != "foo" ? |
| 18:57.39 | p3nguin | Let is say that ${THISISIVR} = lish. |
| 18:57.55 | p3nguin | foo${THISISIVR} then equals foolish |
| 18:58.20 | adeel|work | ah, so another fun way to test if its null |
| 18:58.21 | p3nguin | So the test is just basically checking the THISISIVR variable for null or non-null. |
| 18:58.40 | adeel|work | what's the benefit vs just doing != "" ? |
| 18:58.41 | p3nguin | It's exactly the same as testing "${THISISIVR}" != "" |
| 18:59.03 | p3nguin | I would have done "${THISISIVR}" != "" or foo${THISISIVR} != foo |
| 18:59.42 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 18:59.50 | p3nguin | When a variable is null, you cannot compare it to nothing, because it will break. So you have to compare something+null against something else. |
| 19:00.11 | p3nguin | In the case of something+null, it could be random charaters or quotes. |
| 19:00.21 | adeel|work | p3nguin, so it turns out, it was actually a bug 2 lines up, the similar inbound comparison had the failure clause to the wrong spot |
| 19:00.39 | p3nguin | oh |
| 19:02.30 | adeel|work | yeah, so once i fixed that, the match works as expected |
| 19:03.02 | p3nguin | the original way? |
| 19:03.35 | adeel|work | haven't tried the original way; but i knows yours works |
| 19:03.37 | adeel|work | lets try the original |
| 19:03.57 | kaldemar | func ISNULL, wink wink |
| 19:04.28 | p3nguin | I use ISNULL and EXISTS all the time. |
| 19:04.45 | adeel|work | p3nguin, yup, works both ways |
| 19:04.51 | p3nguin | Nice. Good to know. |
| 19:06.24 | adeel|work | i think yours is more explicit though |
| 19:07.38 | adeel|work | although, i still may need to split this gotoif into a couple gotoif's |
| 19:11.57 | p3nguin | Perhaps instead of a compound expression, use two expressions. |
| 19:13.24 | p3nguin | I don't know if that will have the same possible combinations or not, but I think it might. |
| 19:15.54 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:16.49 | adeel|work | i need to put this dialplan change through its paces....depending on what breaks, i might need to do a test to see if only 1 of the 2 variables are set, and dumb thing like that |
| 19:17.58 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 19:18.35 | p3nguin | That's what the ISNULL and EXISTS functions are for. |
| 19:20.44 | adeel|work | well, it's more of do something else when both at set simultaneously, as it would imply a different call scenario |
| 19:21.05 | adeel|work | er s/at/are/ |
| 19:25.52 | p3nguin | I think, usually, internal vs. external calling is determined by the extension. |
| 19:26.04 | p3nguin | At least that's how I determine where the call goes. |
| 19:27.28 | adeel|work | well what happens when you have a call-forward on busy/unavail, then both flags might be set |
| 19:29.56 | *** join/#asterisk clintc (~clintc@n128-227-81-159.xlate.ufl.edu) |
| 19:36.40 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 19:39.10 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 19:39.46 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:43.18 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:44.57 | *** join/#asterisk hfb (~hfb@pool-96-229-38-102.lsanca.dsl-w.verizon.net) |
| 19:49.20 | p3nguin | adeel|work: My call forwarding works in the same way a direct call works -- destination extension decides where it goes. |
| 19:50.33 | adeel|work | p3nguin, yeah, but in my setup, it's a little more convoluted, and just need to make sure i'm not overlooking something.... |
| 19:53.43 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 19:56.53 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 20:15.00 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 20:25.15 | *** join/#asterisk orev (~orev@c-24-126-141-58.hsd1.ga.comcast.net) |
| 20:26.13 | orev | anyone have luck faxing using a linksys pap2t? |
| 20:30.53 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 20:32.25 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 20:36.37 | LemensTS | hello, i was asking about hints and blf yesterday. Ive been reading non stop for 2 days now, I have come up with this: http://pastebin.com/SLv7ZnTB I am concerned with the part at the end, the part where 802 is 'InUse' and should be 'Idle' |
| 20:37.59 | p3nguin | You can't use variables in hints. |
| 20:38.28 | LemensTS | p3nguin: Do you mean exten = _80[0-8],hint,SIP/${EXTEN} <-- ${EXTEN} |
| 20:38.36 | p3nguin | correct |
| 20:38.40 | p3nguin | And I don't know if patterns work, either. |
| 20:39.36 | p3nguin | You say, "If I call 201..." |
| 20:39.42 | p3nguin | But you do not have a hint for 201. |
| 20:39.54 | p3nguin | No hint for the extension you've dialed, no BLF. |
| 20:40.49 | p3nguin | It is clear to me that you do not understand the separation between an extension and a phone, specifically as hints pertain to it. |
| 20:41.46 | p3nguin | "If I call 201..." hints will match 201,hint,<device> ... which you do not have. |
| 20:42.05 | p3nguin | 201 is the extension. That's what the hint uses. |
| 20:42.24 | p3nguin | Having stupid names for phones makes things a lot more confusing. |
| 20:44.26 | jaytee | do you mean like SIP/GUMP or SIP/LloydChristmas ? |
| 20:44.31 | LemensTS | Ok I understand all that. If I make a '201,hint,SIP/800&SIP/802)' and remove the pattern matching and variable you think 802 will quit showing InUse |
| 20:44.49 | p3nguin | jaytee: No, I mean like SIP/800 vs. extension 800. |
| 20:45.01 | jaytee | I know, was just joking |
| 20:45.04 | p3nguin | oh |
| 20:45.10 | p3nguin | didn't know |
| 20:45.37 | jaytee | well, GUMP from Forrest Gump and Lloyd Christmas (Jim Carey) from Dumb and Dumber. |
| 20:45.49 | p3nguin | I think if you call 201 and SIP/802 picks up the phone, hint 201 is going to report InUse.. because it is IN USE. |
| 20:46.24 | p3nguin | It will continue to be IN USE until it is no longer IN USE. |
| 20:46.36 | p3nguin | At which time it will go Idle again. |
| 20:47.09 | p3nguin | If you call 201 and pick up the phone at SIP/802, go check "core show channels" and tell me if you don't see a call to 201@your-context active. |
| 20:47.36 | p3nguin | When that active channel ends, THEN the hint should report that 201 is no longer InUse. |
| 20:47.51 | jaytee | so hints cannot use variables like in his pastebin? they have to be explicit for extension with sip/devicename |
| 20:47.58 | jaytee | ? |
| 20:47.59 | p3nguin | yes |
| 20:48.01 | p3nguin | that's correct |
| 20:48.19 | jaytee | ok, so using the pattern match would be pointless too then, wouldn't it? |
| 20:48.24 | p3nguin | hints, for a reason I do not know, do not work with variables. |
| 20:48.37 | p3nguin | And I have no idea if pattern matching works or not. |
| 20:48.52 | jaytee | easy enough to test I guess |
| 20:48.55 | p3nguin | But I do think it would be pointless, even if patterns do work. |
| 20:49.23 | jaytee | yeah, cuz if variables won't work then using a pattern match would be a disaster. |
| 20:49.44 | jaytee | or just not work at all |
| 20:49.54 | p3nguin | You would show In Use for things that don't necessarily need to be marked as In Use. |
| 20:49.59 | jaytee | yep |
| 20:50.20 | p3nguin | I hope it won't even work with patterns. That would preempt the disaster. |
| 20:50.37 | jaytee | that's if the you explicitly named the sip/device instead of using sip/${EXTEN} |
| 20:51.19 | jaytee | then any extension that matched the pattern would link to the same device |
| 20:52.20 | beek | waves to jaytee |
| 20:52.29 | jaytee | waves back to beek |
| 20:52.36 | beek | been a while my friend. |
| 20:53.00 | jaytee | beek, how've you been? it has been awhile although I see you in Vyatta from time to time. |
| 20:53.10 | *** join/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 20:53.20 | beek | Doing pretty well... suffering from bronchitis as the moment. How about yourself? |
| 20:53.28 | Steel_Reign | afternoon all |
| 20:53.40 | beek | good afternoon |
| 20:53.43 | LemensTS | p3nguin: If you call 201 and pick up the phone at SIP/802, go check "core show channels" and tell me if you don't see a call to 201@your-context active. <---yes |
| 20:53.51 | jaytee | other than a touch of arthritis I'm doing ok. busier than a one legged man in a butt kickin contest |
| 20:54.03 | p3nguin | Yes you see a call to 201? |
| 20:54.08 | beek | I just hope that you're doing the kicking! |
| 20:54.20 | LemensTS | p3nguin: yep |
| 20:54.24 | p3nguin | Then 201 is InUse. The hint should report is as such. |
| 20:54.26 | jaytee | not always :-( |
| 20:55.33 | Steel_Reign | can anyone tell me why i am seeing this. http://pastebin.com/50AqWiWw |
| 20:55.36 | p3nguin | And if SIP/802 picks up the phone and makes a call to some other destination, the hint should also report extension 201 is In Use because the device related to the hint is being used. |
| 20:55.41 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
| 20:59.26 | Steel_Reign | and this >> http://pastebin.com/WzcWFQzj |
| 20:59.46 | Steel_Reign | any insight would be greatly appreciated |
| 20:59.48 | p3nguin | cannot support a java asterisk manager app. |
| 21:00.40 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pauptktwigzchgsz) |
| 21:01.34 | Steel_Reign | do you know how i can solve that peguin? |
| 21:01.46 | p3nguin | Stop using said app. |
| 21:01.59 | p3nguin | Configure in a sane manner. |
| 21:02.03 | p3nguin | ... |
| 21:02.05 | p3nguin | Profit. |
| 21:03.07 | LemensTS | p3nguin: I appreciate your help, it seems to be working now how it should. I will test in more detail when I get home. It seemed like my main problem was the pattern matching and variable in the hints. |
| 21:03.38 | LemensTS | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions At the bottom of this it says you can do the variable in 1.6.1 and later...rather that is true or not is another story |
| 21:04.08 | LemensTS | It may have just been my pattern matching causing the problem. |
| 21:05.16 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
| 21:07.49 | p3nguin | I'll have to test variables in hints in 1.8. I know it didn't work when I tried it in 1.4. |
| 21:10.09 | p3nguin | In app_chanspy, what is the difference between option E and option S? |
| 21:12.49 | *** part/#asterisk orev (~orev@c-24-126-141-58.hsd1.ga.comcast.net) |
| 21:13.05 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 21:20.02 | _Corey_ | p3nguin: E assumes you're listening on one call/agent and hangs up when their call terminates. S will hang up after you've cycled through your agents/channels when you've used a prefix to match multiple |
| 21:20.13 | Steel_Reign | stopping the use of the app is not an option. the app is openfire |
| 21:20.35 | Steel_Reign | i am trying to get openfire to work through asterisk and bria for iphone |
| 21:20.57 | Steel_Reign | this problem has been driving me nuts for a week now |
| 21:20.58 | _Corey_ | p3nguin: I don't use the "prefix" too much as such, I tend to specify a specific channel |
| 21:21.27 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 21:22.45 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:22.45 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.1 (2012/03/15), 1.8.10.1 (2012/03/15), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:23.38 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 21:24.02 | grandpapadot | Hey guys, is there a way to set a HOLD timeout in asterisk 1.8? i.e., if the call is on hold longer than X seconds, send to <whatever,s,1>? |
| 21:24.51 | Kobaz | nothing built in |
| 21:25.13 | grandpapadot | Kobaz: tnx :) |
| 21:25.18 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 21:26.24 | Kobaz | could do it with the ami if you add a hold event in channel.c |
| 21:28.31 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 21:31.05 | p3nguin | _corey_: I use ChanSpy(,q) to cycle through existing channels and to wait for new ones to be created. |
| 21:31.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:38.59 | _Corey_ | p3nguin: Gotcha... hope that explanation makes sense then |
| 21:39.14 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 21:39.59 | p3nguin | Basically, E and S will do the same thing in that case. |
| 21:40.02 | p3nguin | End. |
| 21:41.08 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 21:41.17 | _Corey_ | Well, I don't know what E would do exactly in your scenario |
| 21:41.32 | _Corey_ | S should do what you'd want I suppose |
| 21:44.00 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
| 21:44.15 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:53.58 | *** part/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 21:54.18 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 22:06.28 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:12.58 | *** join/#asterisk irocklinux|2 (~kvirc@169.151.165.156) |
| 22:20.38 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 22:22.31 | fprior | [30% OT] Hi, anyone worked with huawei dongle with chan_dongle ? My dongle is detected as cdrom. Can you help me to resolve this problem ? |
| 22:49.37 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
| 23:09.22 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-135-101.dsl.stlsmo.sbcglobal.net) |
| 23:15.33 | *** join/#asterisk nW44b_ (~Schnitzel@unaffiliated/benwa) |
| 23:27.28 | *** join/#asterisk irocklinux (~kvirc@cpe-76-183-171-186.tx.res.rr.com) |
| 23:33.21 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a07d3.pool.mediaWays.net) |
| 23:39.50 | *** join/#asterisk timahvo1 (~rogue@41.80.144.67) |
| 23:40.48 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:55.29 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net) |