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| 03:31.45 | pensmit | I deleted the /var/log/asterisk/full file. How do you make it come back? |
| 03:32.28 | WIMPy | 'logger rotate' |
| 03:32.48 | WIMPy | (for example) |
| 03:33.05 | pensmit | well i don't want to rotate them for now |
| 03:33.20 | pensmit | it's just i deleted it and it isn't recreating it |
| 03:33.23 | pensmit | for some reason |
| 03:33.40 | WIMPy | Probably because it's still writing to the old one. |
| 03:33.50 | pensmit | the old one is gone |
| 03:33.51 | pensmit | rm |
| 03:33.53 | pensmit | rm full |
| 03:34.07 | WIMPy | That's not the way unix works. |
| 03:34.19 | pensmit | right right |
| 03:34.28 | WIMPy | You can stil write to and read from deleted files. |
| 03:34.36 | pensmit | lol |
| 03:35.00 | WIMPy | seriousely |
| 03:35.12 | pensmit | never heard of it |
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| 03:35.16 | pensmit | got a link |
| 03:35.43 | WIMPy | NFI where you can read about it. Maybe in man rm. |
| 03:35.54 | pensmit | unless in the programming it is pointing to an actual inode value |
| 03:35.58 | pensmit | i seriously doubt this |
| 03:36.04 | WIMPy | A file won't get deleted while it's in use. Only the name is removed. |
| 03:36.07 | ChannelZ | Files are kind of 2 pieces.. the blocks on disk and the entries that refer to them in the filesystem. |
| 03:36.09 | pensmit | now how do i get the software to recrate the file |
| 03:36.44 | pensmit | hmm |
| 03:36.46 | ChannelZ | you don't |
| 03:36.54 | ChannelZ | I mean not what's been written afterwards. |
| 03:36.54 | pensmit | guys |
| 03:36.56 | pensmit | come on man |
| 03:37.00 | pensmit | give me the answer here |
| 03:37.03 | ChannelZ | You need to get the app to close the filehandle it's writing to and open a new one. |
| 03:37.07 | pensmit | i feel like you're fucking with me and im worried |
| 03:37.09 | WIMPy | You can always restart Asterisk if you don;t want to rotate. |
| 03:37.17 | ChannelZ | Usually you signal it with an alarm |
| 03:37.36 | pensmit | ok i deleted full.0 full.1 |
| 03:37.40 | pensmit | and full.2 and full |
| 03:37.46 | pensmit | I assumed it was done with the others |
| 03:37.54 | pensmit | but currently writing to full |
| 03:38.04 | pensmit | so what you wrote makes sense |
| 03:38.25 | pensmit | I can't restart asterisk right now |
| 03:38.30 | ChannelZ | you don't have to |
| 03:38.33 | pensmit | so i'm screwed? |
| 03:38.43 | ChannelZ | I'm not sure really what your question is now |
| 03:38.52 | ChannelZ | Is it writing to the log or not? |
| 03:39.01 | pensmit | I deleted full |
| 03:39.04 | pensmit | you tell me |
| 03:39.06 | WIMPy | It's writing to the deleted file. |
| 03:39.10 | pensmit | ok |
| 03:39.12 | ChannelZ | restart logger |
| 03:39.25 | pensmit | the asterisk logger? |
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| 03:39.29 | ChannelZ | yes |
| 03:39.37 | pensmit | ok..new to this |
| 03:39.38 | pensmit | how |
| 03:39.45 | *** part/#asterisk Sha`Bren (~Sha`Bren@pdpc/supporter/active/shabren) |
| 03:39.52 | ChannelZ | "logger reload" on the console |
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| 03:40.04 | pensmit | and that shouldn't harm anything else? |
| 03:40.10 | ChannelZ | no |
| 03:40.14 | pensmit | ok |
| 03:40.15 | pensmit | hey |
| 03:40.16 | pensmit | thanks guys |
| 03:40.20 | pensmit | i did not know that |
| 03:40.30 | pensmit | it continued writing once the file was deleted |
| 03:40.49 | pensmit | sometimes you worry about who to trust to...you know |
| 03:40.50 | ChannelZ | if you look at logrotate scripts for many other apps you will see they do the same thing. |
| 03:40.50 | pensmit | so thanks |
| 03:40.59 | pensmit | ok thanks a lot |
| 03:41.03 | pensmit | thanks WIMPY |
| 03:41.08 | ChannelZ | IE you need to signal the process to tell it to close its filehandles and open new ones. |
| 03:42.10 | pensmit | hey guys |
| 03:42.14 | pensmit | guess what |
| 03:42.17 | pensmit | I did it and it worked |
| 03:42.24 | pensmit | I owe you each 2 Internet beers |
| 03:42.28 | pensmit | :) |
| 03:42.40 | WIMPy | Ok, so now we know that a logger reload will reopen the log files. |
| 03:43.07 | pensmit | lmao |
| 03:43.26 | pensmit | that really damn tickled me |
| 03:43.26 | WIMPy | Well, I never had to try. |
| 03:45.24 | pensmit | You should have left it with your first line there |
| 03:45.26 | pensmit | funny as hell |
| 03:45.52 | pensmit | read it from my perspective with me assuming you're a superguru |
| 03:46.11 | WIMPy | You know that by "reopen" I mean re-open as in close and open, nicht as in recreate? |
| 03:47.22 | pensmit | Don't be funny with a confused man |
| 03:47.26 | pensmit | I know that |
| 03:47.32 | pensmit | i deleted the file pointer |
| 03:47.34 | pensmit | not the file |
| 03:47.47 | pensmit | the file was still open by the logger and it just reopened it but recreated the pointer? |
| 03:47.49 | pensmit | correct? |
| 03:48.14 | WIMPy | No. At the time the last process closes the file, it is really deleted. |
| 03:48.24 | pensmit | ahh I see that now |
| 03:48.30 | pensmit | because the file is much smaller |
| 03:48.31 | WIMPy | With large files you can see that on your disk usage. |
| 03:48.36 | pensmit | yes |
| 03:48.45 | pensmit | so it didn't reopen it |
| 03:49.04 | WIMPy | And then when it wanted to open the file to append to it, it wasn't there, so it created a new one. |
| 03:49.55 | pensmit | it could no longer get the inode location? |
| 03:50.15 | pensmit | it had it when it still had the file opened? |
| 03:50.19 | WIMPy | It really didn;t exist any more at that point. |
| 03:50.35 | pensmit | well it's still truly on the hard drive |
| 03:50.43 | pensmit | just the pointers to it are gone |
| 03:50.55 | WIMPy | Yes, but there's no reference to it any more. |
| 03:50.59 | pensmit | yes |
| 03:50.59 | pensmit | ok |
| 03:51.02 | pensmit | good enough |
| 03:51.06 | pensmit | i'm tired...long day |
| 03:51.15 | pensmit | thanks...i know there is more i don't know at the least |
| 03:51.18 | pensmit | thanks a lot |
| 03:51.54 | *** part/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 03:51.55 | WIMPy | And if you had any process running that analyzes the logfile, like fail2ban, it would still have worked after you "deleted" the file. |
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| 06:21.24 | ChannelZ | wtf |
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| 06:33.43 | ChannelZ | ok.. so I have a file sitting on my Windoze box which is mounted on my Linux box via CIFS. From the linux box I can see it. I can cat it. If I try to pico it though, it tells me 'New file' |
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| 06:49.04 | blehxor | I've set storesipcause=yes |
| 06:49.16 | blehxor | but sip show settings says its disabled still |
| 06:49.48 | blehxor | I read some forum post claiming it took an hour to go into effect? anyone else experience this? |
| 06:49.56 | blehxor | I've restarted asterisk |
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| 06:59.58 | kaldemar | blehxor: the option was deprecated in 11 anyway. |
| 07:00.15 | ChannelZ | huh. never even heard of it. |
| 07:01.43 | kaldemar | it can be used to disable cause sets by channels to the SIP_CAUSE hash on every response. |
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| 07:08.59 | blehxor | hm |
| 07:09.24 | blehxor | so you always get congested on call setup failure? |
| 07:09.30 | blehxor | no way around that? |
| 07:11.14 | blehxor | i want to get a real status/response message when my SIP call fails... :( |
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| 07:35.14 | kaldemar | blehxor: use DIALSTATUS and HANGUPCAUSE() after your Dial. |
| 07:35.40 | phix | nice |
| 07:36.31 | blehxor | When I try those I get congested |
| 07:36.50 | blehxor | "congested" or code 34 for hangupcause, which maps to 34 |
| 07:37.00 | blehxor | er congested |
| 07:37.01 | blehxor | tired |
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| 09:02.15 | nam3l3zz | hello |
| 09:03.44 | nam3l3zz | quick one, i am connected to a DID provider via sip, i have multiple DID numbers via same sip connection/extension, {EXTEN} returns extension rather then DID number dialed, how do I extract the number dialed ? |
| 09:03.57 | *** part/#asterisk fling (~fling@fsf/member/fling) |
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| 09:10.24 | kaldemar | nam3l3zz: try ${CALLERID(dnid)} |
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| 09:48.25 | nam3l3zz | kaldemar: will do, thanx |
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| 09:52.56 | zeves | ADMIN |
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| 09:56.54 | nam3l3zz | kaldemar: gives extension/sip username, rather DID number :? |
| 09:57.16 | nam3l3zz | kaldemar: can it be something wrong in sip.conf or did provider side ? |
| 09:59.12 | kaldemar | nam3l3zz: may be either. how did you set it up? |
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| 10:00.49 | nam3l3zz | kaldemar: http://pastebin.com/MmwjG1gg |
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| 10:07.48 | kaldemar | nam3l3zz: your register line tells them to send the call to the number that you have after "/". |
| 10:08.48 | kaldemar | look at sip debug in different cases, do you see what you expect in the invites? |
| 10:51.50 | nam3l3zz | kaldemar: will try to change stuff after forwardslash, will look at invites with commview sniffer/decoder |
| 10:51.58 | nam3l3zz | *thnx |
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| 11:01.23 | kaldemar | nam3l3zz: you can use asterisk. "sip set debug on" |
| 11:10.37 | nam3l3zz | kaldemar: i'v found via commview, in the header, X-voipnow-did being set to the right dst number |
| 11:12.13 | kaldemar | you'll get it in dialplan with func SIP_HEADER |
| 11:13.18 | nam3l3zz | ${SIP_HEADER(X-voipnow-did)} |
| 11:14.28 | nam3l3zz | kaldemar: u'r one of the best in here, in the sense of right attitude, thnx again |
| 11:20.41 | kaldemar | np |
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| 11:32.04 | bdar77 | guyz i am stuck since yesterday |
| 11:32.05 | bdar77 | rtp_engine.c: No RTP engine was found. Do you have one loaded? |
| 11:34.16 | bdar77 | its showing XXX in menuselect for res_rtp_asterisk |
| 11:34.55 | acidfu | you need to install the dependency |
| 11:35.37 | bdar77 | i did install uuid |
| 11:35.45 | acidfu | then redo ./configure |
| 11:35.49 | bdar77 | let me restart the box and c |
| 11:35.57 | acidfu | restart the box ? |
| 11:36.13 | acidfu | and write properly |
| 11:36.24 | bdar77 | okay |
| 11:36.57 | kaldemar | don't restart your box, it's a waste of time. |
| 11:40.50 | nam3l3zz | kaldemar: can I reject a sip invite from a dialplan ? |
| 11:40.52 | bdar77 | It's my VM, giving me a hard time too |
| 11:45.52 | Greenlight | bdar77: This asterisk 11.5? |
| 11:46.44 | bdar77 | yes |
| 11:46.57 | bdar77 | got the module working |
| 11:47.21 | bdar77 | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 11:47.27 | bdar77 | getting this message again |
| 11:47.31 | Greenlight | Asterisk isn't running |
| 11:47.36 | Greenlight | Did you start it? |
| 11:47.38 | bdar77 | fixed it yesterday |
| 11:47.46 | bdar77 | it's running |
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| 11:53.15 | bdar77 | i used asterisk -c to get over asterisk.ctl issue |
| 11:54.32 | Greenlight | You mean, you started it. |
| 12:02.18 | bdar77 | yeah |
| 12:02.48 | bdar77 | ls /usr/lib/asterisk/modules/ | grep rtp |
| 12:02.48 | bdar77 | chan_multicast_rtp.so |
| 12:02.48 | bdar77 | res_rtp_multicast.so |
| 12:04.40 | bdar77 | res_rtp_asterisk Depends on: uuid(E) |
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| 12:06.51 | bdar77 | rpm -qa | grep uuid |
| 12:06.51 | bdar77 | libuuid-2.17.2-12.9.el6_4.3.x86_64 |
| 12:06.51 | bdar77 | uuid-devel-1.6.1-10.el6.x86_64 |
| 12:06.51 | bdar77 | uuid-1.6.1-10.el6.x86_64 |
| 12:06.51 | bdar77 | uuid-c++-1.6.1-10.el6.x86_64 |
| 12:06.51 | bdar77 | libuuid-devel-2.17.2-12.9.el6_4.3.x86_64 |
| 12:11.28 | Greenlight | Is there a particular question you're asking ? |
| 12:12.00 | bdar77 | i cant load res_rtp_asterisk module |
| 12:12.11 | Greenlight | DOes it give an error ? |
| 12:12.17 | bdar77 | nope |
| 12:12.31 | bdar77 | how to load it? |
| 12:12.46 | Greenlight | [01:04pm] <bdar77> res_rtp_asterisk Depends on: uuid(E) <-- ? |
| 12:13.02 | Greenlight | Doesn't seem like the module is there |
| 12:13.27 | Greenlight | Have you run ./configure, and then make && make install |
| 12:16.29 | bdar77 | yes, let me try once more |
| 12:30.54 | bdar77 | <Greenlight> it's working now |
| 12:30.54 | kesselklopfer79 | you can also ./configure and then make menuconfig and look for the things you need .... so you can see it will be compiled |
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| 12:39.01 | bdar77 | Yipee, got my first asterisk softphone to softphone call working :) |
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| 13:05.58 | AL13N | at the end of the call, i would like the dialplan to make a second call to another person, play a message and then hangup, i got everything set; but after the 1st hangup nothing more gets executed |
| 13:06.15 | AL13N | the second call is never made |
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| 13:32.55 | WIMPy | AL13NL: Did you tell Dial() to continue after the call has ended? |
| 13:33.43 | jmetro | with a ,g i believe. |
| 13:44.45 | *** join/#asterisk serafie (~erin@nat/digium/x-eiwkuzjtfdhehnpr) |
| 13:52.35 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 13:57.23 | igcewieling | Actually I think you tell dial to continue after the call ends by typing "core show application dial" in the Asterisk CLI |
| 14:02.06 | jmetro | that would work too |
| 14:04.08 | *** join/#asterisk Dovid (~Dovid@96.sub-70-208-75.myvzw.com) |
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| 14:15.49 | Katty | QWELL |
| 14:15.57 | Katty | WHAT DID YOU DO |
| 14:16.52 | Katty | Qwell: congratulations, sir!!!! |
| 14:30.17 | jmetro | Katty: he won the marathon? |
| 14:30.49 | Katty | even better. |
| 14:30.55 | Katty | he proposed. |
| 14:30.58 | Katty | and didn't get turned down! |
| 14:31.06 | jmetro | Oh, the longest marathon :3 |
| 14:31.16 | Katty | ^_____^ |
| 14:39.24 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 14:40.19 | SuperNull | does changing compat section required a hard reload ? (specifically pbx_realtime from 1.6->1.4) |
| 14:42.12 | igcewieling | marriage? My condolences. |
| 14:42.25 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 14:42.26 | igcewieling | SuperNull: I assume it does. |
| 14:44.17 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 14:46.09 | igcewieling | SuperNull: I've never seen anything in asterisk.conf work with just a reload. |
| 14:46.17 | SuperNull | alright. that is good news. |
| 14:46.58 | wasanzy | when I run this: snmpwalk -v 3 -u asteriskZabbix -n "" -l authPriv -a MD5 -A astrisk123 -X astrisk123 localhost asterisk |
| 14:47.12 | wasanzy | I am getting the error: asterisk: Unknown Object Identifier (Sub-id not found: (top) -> asterisk) |
| 14:48.58 | SuperNull | do you have the mib loaded ? |
| 14:49.18 | wasanzy | SuperNull: me? |
| 14:49.18 | SuperNull | and is 'asterisk' the root of the mib ? |
| 14:49.21 | SuperNull | yeah |
| 14:49.42 | wasanzy | SuperNull: I didn't do that |
| 14:49.44 | SuperNull | also does it have support for v2c ? |
| 14:50.08 | SuperNull | never really played with v3 snmp. seems overly complex |
| 14:51.28 | wasanzy | SuperNull: I don't know if it has support for v2 |
| 14:52.12 | SuperNull | you using res_snmp i assume ?.. |
| 14:52.50 | SuperNull | you might make me try to get it working ;) i want some stats from my boxes to but i dont even know what res_snmp reveals that could be useful. |
| 14:52.54 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:55.35 | SuperNull | igcewieling i think the 1.4 compat mode just does a hard replace on '|' to ',' right? (im seeing commas in the active dialplan) |
| 14:56.08 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:56.09 | igcewieling | SuperNull: no idea. only crazy people want 1.4 compat 8-| |
| 14:56.24 | SuperNull | ... i know. |
| 14:56.27 | SuperNull | only in realtime tho.. |
| 14:56.51 | SuperNull | the way they previously wrote this was a megaton of 'hard code' macros .. and using realtime extension table as a method to proxy it to the correct macro. |
| 14:56.52 | SuperNull | its so bad. |
| 14:57.16 | SuperNull | my goal is to not use realtime extensions at all.. because i dont see a use for it. |
| 14:57.30 | igcewieling | *nod* |
| 14:57.36 | SuperNull | o/ |
| 14:58.30 | igcewieling | We keep our peers in a "realtime" sippeers table, but don't use realtime. We export the sippeers to sip.conf. |
| 14:58.32 | jmetro | Realtime is pretty awesome. |
| 14:58.54 | igcewieling | We used Realtime for a short while. I consider it one of Asterisk's blue bells features. |
| 14:59.26 | jmetro | I still dont know what issues you see with it. It works great. |
| 14:59.57 | igcewieling | jmetro: for one thing qualify doesn't work right across multiple server |
| 15:00.15 | SuperNull | it works fine for registration .. we had some odd issue tho.. where when you had a couple hundred registrations it would force asterisk into completly no response.. only happened with rtcachefriends=yes and rtautoclear=yes |
| 15:00.15 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:00.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:00.35 | igcewieling | You can't separate out the database reads from the database writes, so you can't have a locally replicated DB for realtime reads and a central master DB for the writes. |
| 15:00.43 | igcewieling | SuperNull: we don't use registration |
| 15:00.50 | SuperNull | gotcha. |
| 15:01.10 | SuperNull | i do. we get hacked otherwise.. anytime the bossman puts in custom sip.conf config.. we got nailed for like 10k in international calls. |
| 15:01.22 | igcewieling | If we did then we would have three servers fighting over the same entries as the servers each update the same item as the same peer registers to multiple servers |
| 15:01.36 | SuperNull | i force even 20 line 'trunks' to register all lines heh. |
| 15:01.41 | igcewieling | SuperNull: 100% of our clients are on static IPs |
| 15:01.58 | SuperNull | <-- small cable operator |
| 15:02.08 | SuperNull | so most of it is residential dynamic ip |
| 15:02.40 | igcewieling | We only offer business class service, so none of this residential sillyness |
| 15:02.51 | SuperNull | yeah. |
| 15:03.14 | igcewieling | If you are stupid enough to not use us for your internet, we still require a static IP to connect to our SIP service. |
| 15:03.33 | SuperNull | worst is setting up an uncapped link to someone running an open source box like FreePBX .. ownage every time. people have no clue how had SIP 'cracking' is these days. |
| 15:03.53 | igcewieling | SuperNull: the issue with FreePBX is the GUI -- it isn't secure |
| 15:03.55 | SuperNull | 30k in international calls.. from a server at russia telekom |
| 15:04.43 | igcewieling | SuperNull: we have three systems which try to control fraud. We have our draconion firewall settings, an internal fraud control script, and the carriers will notify and block if they see odd traffic. |
| 15:04.47 | *** join/#asterisk tapout_ (~tapout@unaffiliated/tapout) |
| 15:04.55 | SuperNull | i have seen to many exploits through apache/php combo. more so tho on highly used deployable packages .. not custom stuff. |
| 15:05.04 | SuperNull | we have a hard limit set for international that is very low.. |
| 15:05.36 | igcewieling | each customer has a set limit per month |
| 15:06.07 | SuperNull | how do you end up verifying that .. do you look in sql cdrs and count up the billsec for 011/11 calls ? |
| 15:06.34 | SuperNull | i think im going to implement that.. i like the idea. |
| 15:09.36 | igcewieling | SuperNull: we hired an external programmer. I believe at the start of an international call it checks the DB for number of remaining mins and lets the call through if there are more than X mins. At the end of the call the database is updated with the mins used for the call. |
| 15:09.52 | SuperNull | hmm nice. |
| 15:10.34 | igcewieling | from MY point of view, we run the script, it sets a dialplan variable and we decide what to do based on that |
| 15:10.51 | jmetro | seems like that should have been an easy script |
| 15:11.15 | igcewieling | SuperNull: we get a device hacked about once a month. |
| 15:13.11 | *** join/#asterisk eduzimrs (~eduzimrs@extranet.explorercallcenter.com.br) |
| 15:14.40 | SuperNull | igcwieling i think i wouldn't use my CDR table.. latency on response would be too high.. i would have to make a cumulative international call time table, insert call time on call completion, check for used call time on dial |
| 15:15.15 | igcewieling | SuperNull: our antifraud script does not use the cdr table for those rerasons |
| 15:15.46 | igcewieling | We also don't worry too much about accurate CDRs because we bill based on the Carrier CDRs not our own Asterisk CDRs |
| 15:16.04 | SuperNull | yeah.. to much latency. 50 million call records since january.. so that could take a bit. |
| 15:16.30 | SuperNull | ehhh we should .. i have NEVER seen a CDR CSV from our providers.. not sure why.. |
| 15:17.04 | SuperNull | also since we do caller id blocking they cannot provide everything without at least special identification or something. so .. either way. |
| 15:19.24 | igcewieling | heh, if our carrier can't find a valid TN to bill to the call gets rejected |
| 15:19.42 | igcewieling | Well, Verizon. Level3 will bill those to a "default" TN. |
| 15:21.20 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:ac78:d36a:e6dc:5da2) |
| 15:21.36 | SuperNull | yeah.. we had L3, we also have a 'sister company' that is a full CLEC with DMS we pretty much do everything with them now. |
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| 15:27.18 | jmetro | I just broke my eyes or something.. If i dont come back tomorrow its because i went blind. |
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| 16:26.40 | SGjunior | Hey guys, I have been going around asterisk for a few months already but I still have some trouble with some key concepts. Although the system I programmed is about to go live, I am still looking to make it more efficient. Incoming calls are quite easy and so is originating an outgoing. What I find hard is handling the calls while they are connected. For outgoing, I create a call file in VFP |
| 16:26.40 | SGjunior | and trop in in /.../outgoing, * dials a phone, I answer(for now since I want to implement SIPAddHeader() later on for auto-answer) and the it dials out through a dialplan extension to where I want (getting the where I want from MySQL). I am looking for a way to transfer calls without having to use the transfer button on my hardphones just like I employ a script from VFP to make outgoing calls. |
| 16:26.40 | SGjunior | Would anyone have a recommendation for this? |
| 16:28.20 | igcewieling | I'm not going to try and parse all that, but I can tell you that if you want to do "stuff" before dialing out, use Local/ channels in your spool file. |
| 16:31.01 | igcewieling | If you want to control calls from outside of Asterisk look at AMI / Manager interface |
| 16:31.16 | igcewieling | You can also generate calls using AMI and Originate rather than spool files. |
| 16:33.35 | SGjunior | fair enough, how about this then; I want to transfer a call to an inactive channel? |
| 16:33.53 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 16:34.35 | igcewieling | Huh |
| 16:35.12 | igcewieling | If you want to transfer a call you can do so via AMI / Manager. The Redirect command, I think, but I've not looked at it in a while. |
| 16:35.53 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 16:38.04 | SGjunior | I can redirect an active channel with the AMI to an app or an ext no problem. would it be using redirect to app Dial(where you want to dial) |
| 16:39.08 | SGjunior | that would be unattended and the information about the channel you want to redirect would have to be stored somewhere |
| 16:41.25 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 16:41.50 | igcewieling | Why not use FOP2? |
| 16:45.34 | SGjunior | looking into it |
| 16:45.40 | SGjunior | thanks! |
| 16:46.55 | navaismo | better and free Monast |
| 16:47.09 | navaismo | IMO^ |
| 16:47.55 | SGjunior | Monast > FOP2? |
| 16:48.34 | navaismo | its like the same thing different look but monast doesnt have a cost |
| 16:48.55 | igcewieling | Is there a Monast module for FreePBX? |
| 16:52.05 | pigpen | anybody around that knows more advanced isc dhcp config using vendor classes? |
| 16:52.17 | pigpen | I guess I should say, hardware classes. |
| 16:52.58 | pigpen | Not exactly "asterisk" but it is for ip address assignment to polycom, cyberdata, and such |
| 16:58.10 | navaismo | igcewieling, nope but you can install it anyway |
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| 17:44.33 | igcewieling | jmetro: Braille IRC? |
| 17:44.42 | jmetro | nah it was something Supernull put up earlier |
| 17:47.53 | *** join/#asterisk monsterco (4c40b682@gateway/web/freenode/ip.76.64.182.130) |
| 17:49.12 | monsterco | Hi everyone - can I have someone explain to me how UDP ports are opened after SIP signals passed authentication? I have a one way audio issue and trying to resolve it. I want to know specifically if my Asterisk attempts to open UDP ports for RTP or the other party first tries to connect to my Asterisk server for an outgoing call from Asterisk? |
| 17:49.43 | monsterco | I am using virtual IPs on pfSense which is set to local Asterisk LAN client |
| 17:52.37 | *** join/#asterisk serafie (~erin@nat/digium/x-xusqktbuprizdfwk) |
| 17:52.59 | *** join/#asterisk Alex_Bkash (b4ea1742@gateway/web/freenode/ip.180.234.23.66) |
| 17:53.30 | Katty | SO. |
| 17:53.45 | jmetro | Katty: Good news, I'm not blind. |
| 17:53.49 | Katty | YAY! |
| 17:53.54 | Katty | not blind is always amazing. |
| 17:54.05 | Katty | also, was there potential for blindness? |
| 17:54.26 | jmetro | I feel like there was, it was strange. But im normal again. |
| 17:54.35 | Alex_Bkash | asterisk calling card and pin less calling how can be implemented? |
| 17:54.39 | Katty | i'm very glad to hear that you're feelign normal in the vision department :> |
| 17:54.46 | Katty | SO. |
| 17:54.52 | Katty | let's say i'm 100. |
| 17:55.01 | Katty | and i decide to dial mister jmetro at 200. |
| 17:55.18 | Penguin | alex_bkash: How will you authenticate the caller? Phone number only? Voice recognition? |
| 17:55.19 | Katty | i get his voicemail and press *, deciding to check my own voicemail, rather than leaving him a message. |
| 17:55.40 | igcewieling | monsterco: your best option is to read the RFC. At least one end of each of the two UDP/RTP streams will use a port assigned by Asterisk |
| 17:55.45 | Alex_Bkash | phone |
| 17:55.52 | *** join/#asterisk jsjc (~Adium@39.Red-2-136-115.dynamicIP.rima-tde.net) |
| 17:55.52 | Penguin | presses katty's button |
| 17:55.53 | Alex_Bkash | with did |
| 17:55.57 | Katty | is there a way to carry my extension 100, over to the a,1,VoicemailMain and dump it in there? |
| 17:56.15 | Katty | regardless of the fact that i dialed 102 |
| 17:56.45 | Penguin | If you use caller ID and if your caller ID shows 100 for your phone number, maybe it will be possible. |
| 17:56.51 | Katty | i'm sure there's a variable for me, who called. but i'm not sure how to pass that over to a,1,Voicemailmain(${something} |
| 17:57.04 | Katty | oooh, callerid. that's clever |
| 17:57.08 | igcewieling | ${CALLERID(num)} |
| 17:57.46 | Katty | what happens if i'm calling in with my mobile |
| 17:57.58 | Katty | and i get transfered to my extension, and press * and it tries to read my mobile callerid |
| 17:58.07 | Katty | i just goes herpderp couldn't read username? |
| 17:58.17 | Penguin | That's exactly what would happen. |
| 17:58.23 | Katty | eggggcelent |
| 17:58.28 | monsterco | igcewieling - I have previously read SIP packets but I tend to forget after a year and then again get confused. So, right now my issue is that I can't hear the other side audio and they can hear me. In firewall logs I see that I Asterisk is attempting to connect UDP but no response from other side - given the virtual IP setup I think it's an issue with pfsense.........any ideas are appreciated |
| 17:58.31 | Penguin | But... you can always rewrite callerid based on your mobile number. |
| 17:58.41 | Katty | oh. my. |
| 17:58.42 | Katty | yes, yes i could |
| 17:58.43 | Penguin | If mobile, change to 100. |
| 17:59.06 | Katty | that seems like a lot of work, but also very clever. |
| 17:59.40 | Alex_Bkash | i need someone to implement that for me :) |
| 17:59.44 | igcewieling | Often when a call comes in I set things up to save the dialed number and the callerid into channel variables so I KNOW I have the original info later in the dialplan |
| 18:00.15 | igcewieling | monsterco: turn off your firwall and see if it starts working. |
| 18:00.31 | igcewieling | Often audio issues are really the call reinviting when it can't reinvite |
| 18:00.47 | Katty | yeah variables are good, but don't always transfer around well |
| 18:00.48 | monsterco | oh, yes, then it works. Even with Automatic NAT on pfsense it works fine - but I turned on Manual Outbound NAT - are you familiar with pfsense? |
| 18:00.53 | Katty | i guess global variables are more handy with that |
| 18:00.54 | igcewieling | monsterco: allow udp port 10,000 - udp port 20,000 as the source port or the destination port and you should be good. |
| 18:01.02 | igcewieling | NAT? NAT??????? |
| 18:01.22 | igcewieling | monsterco: no I'm not familiar with pfsense |
| 18:01.39 | monsterco | yes, 10000-20000 are what Asterisk uses but for outbound call there is no need to open those on Firewall |
| 18:01.46 | monsterco | am I not right with that? |
| 18:01.52 | Katty | masquereds with igcewieling |
| 18:01.56 | [TK]D-Fender | that is INBOUND |
| 18:02.01 | monsterco | because Asterisk will tell the other party to come back on such and such port and it will have it open |
| 18:02.06 | monsterco | I am only making outbound calls |
| 18:02.12 | [TK]D-Fender | asterisk sends out to the port the CLIENT requestes |
| 18:02.13 | Katty | monsterco: audio comes back to you, sweet pea |
| 18:02.25 | igcewieling | If you are behind NAT you want to use Asteirsk's NAT traversal, and NOT any firewall SIP/RTP traversal things |
| 18:02.26 | Katty | monsterco: i can't go BLAHBLAHBLAH to you, if you're blocking me to you |
| 18:02.26 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 18:02.27 | [TK]D-Fender | monsterco: audio goes BOTH ways |
| 18:02.36 | [TK]D-Fender | monsterco: YOUR ports... THEIR port |
| 18:02.42 | Katty | HE SHE OR ITS PORTS |
| 18:02.52 | Katty | welcome to /r/grammar today |
| 18:03.02 | igcewieling | not their ports? |
| 18:03.04 | Katty | brough to you by the letter C and [TK]D-Fender |
| 18:03.27 | monsterco | I thought once Asterisk initiates the first UDP packet so the port opens automatically since it is the initiator of the call. Isn't that true? |
| 18:04.11 | igcewieling | monsterco: correct |
| 18:04.34 | igcewieling | But if Asterisk is behind NAT you need to forward 5060 to the server, set up externip and localnet and turn off direct media |
| 18:04.44 | monsterco | igcewieling - so, then Firewall doesn't have to have port forward inbound enabled for 10000-20000 while only outbound calls are made from Asterisk |
| 18:05.14 | igcewieling | monsterco: in theory no, in practice seldom |
| 18:05.15 | monsterco | I didn't setup externip and localnet just fine - and I did turn on NAT = yes and canrenvite to NO |
| 18:05.32 | igcewieling | without externip asterisk won't know what IP to present the packets from |
| 18:05.44 | igcewieling | and without localnet Asterisk won't know what connections to do NAT fixups for. |
| 18:05.56 | monsterco | igcewieling - but there shouldn't be any port forward 5060 needed either since my outgoing port will be decided by router as I am only making outbound call |
| 18:06.51 | igcewieling | monsterco: if you never receive calls and have no external devices register to you then you likely won't need to port forward 5060, but you still need localnet and externip set |
| 18:07.19 | igcewieling | none of this matters of your firewall is trying to do it's own SIP ALG or SIP SPI |
| 18:07.28 | igcewieling | that will make it not work correctly in Asterisk |
| 18:07.59 | Katty | is there some fancy word for when you press # to skip all the annoying blahblahblah bits? |
| 18:08.14 | monsterco | igcewieling - yes, I am familiar with SIP ALG. pfsense handles that well (at least in automatic outbound nat situation) - I have to see what the virtual IPs are doing |
| 18:08.31 | monsterco | Katty - "to skip this message" |
| 18:08.45 | Katty | err no |
| 18:09.00 | Katty | i mean like a real word. |
| 18:09.05 | igcewieling | If it was working well you would not be here asking questions. |
| 18:09.07 | Katty | a buzz word |
| 18:09.23 | Katty | a buzz word isn't a real word. |
| 18:09.44 | igcewieling | you can use your firewall's built in SIP ALG, in which case go talk to the pfsense people, or you can disable it and enable Asteirsk's NAT traversal. I've never heard of it working when both are enabled. |
| 18:09.48 | Katty | a non-real buzz word :P |
| 18:10.34 | monsterco | igcewieling - I am looking for technical theory |
| 18:12.13 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 18:16.57 | Katty | Alex_Bkash: please don't send me private messages without asking, that's rude. |
| 18:20.41 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:21.16 | Alex_Bkash | i did not mean to be rude |
| 18:21.19 | Alex_Bkash | :( |
| 18:21.25 | wdoekes | but now you know |
| 18:22.35 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:22.57 | Alex_Bkash | Katty:can i ? |
| 18:23.36 | BlackBishop | :-s |
| 18:24.24 | jmetro | Katty> Intro Skip maybe |
| 18:24.35 | jmetro | Katty> It has to include the word Skip, i know that much |
| 18:24.49 | jmetro | For a buzzword, my best shot is "Instructiskip" |
| 18:28.34 | *** join/#asterisk citrusfizz (~chatzilla@70.184.40.66) |
| 18:29.02 | citrusfizz | looking at voip.ms or flowroute for our small office voip system, should i lean toward one or the other? |
| 18:29.22 | jmetro | i remember hearing good things about flowroute, but i dont know myself. |
| 18:29.25 | Penguin | They are both good. |
| 18:29.54 | Penguin | VoIP.ms has a lot of PoP locations. That could be beneficial. |
| 18:30.01 | Alex_Bkash | Hello Penguin i did send you a msg in private |
| 18:30.12 | Penguin | Your mistake. |
| 18:30.37 | Alex_Bkash | i dont mean to be rude Penguin |
| 18:30.52 | carrar | why not |
| 18:31.10 | *** join/#asterisk bdar77 (~bad53@2.90.75.61) |
| 18:31.14 | citrusfizz | we currently use voip.ms for other stuff, and i just had a conversation about upping our concurrent calls, and the tech acted like it was going to be a big deal and the accounting team had to review our logs or something... |
| 18:31.28 | citrusfizz | from 5 to 20.... OK buddy.... |
| 18:31.39 | carrar | hahah |
| 18:31.49 | carrar | "Where not sure we want more biz" |
| 18:32.26 | jmetro | "woah woah woah, 15 calls is a lot, we might have to get a second T1" |
| 18:32.30 | carrar | 5 is good, but man 20!! Thats just too much talking! |
| 18:32.41 | citrusfizz | seriously, really just put me off, and now i have to wait for a ticket to be processed... |
| 18:33.02 | jmetro | theyre trying to think how they can afford to pay the guy to lay an extra cat5 from your office to the modem |
| 18:33.05 | *** join/#asterisk SGjunior (~sgjunior@out-pq-233.wireless.telus.com) |
| 18:33.18 | *** part/#asterisk monsterco (4c40b682@gateway/web/freenode/ip.76.64.182.130) |
| 18:33.25 | carrar | thats 15 more pots lines they need to order |
| 18:34.43 | citrusfizz | haha |
| 18:35.28 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:36.46 | Katty | exten => 1000,1,GotoIf($["{callerid(num)}" = "1000"]?vm) would go to exten => 1000(vm),DoStuff()? |
| 18:37.10 | carrar | O |
| 18:37.11 | carrar | M |
| 18:37.11 | carrar | G |
| 18:37.18 | Katty | hello cupcake! |
| 18:37.20 | Katty | hugs carrar |
| 18:37.23 | carrar | hello sugar pie |
| 18:37.30 | carrar | hugs Katty back! |
| 18:37.32 | Katty | how're you and the lady? |
| 18:37.41 | carrar | we're doing grand |
| 18:37.48 | Katty | egggcellent |
| 18:37.51 | carrar | going to go look at houses |
| 18:38.04 | citrusfizz | as a backup plan, is there a way to queue call files? i have my nagios server send out 6 calls, but one obviously fails. |
| 18:38.22 | citrusfizz | i guess i could write some more logic in my script |
| 18:38.27 | citrusfizz | that creates the call files |
| 18:38.27 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:38.40 | Katty | carrar: horay for house hunting!!! |
| 18:38.47 | carrar | Sound Transit wants to put a train through our house and call it eminent domain, so we have to move |
| 18:38.56 | Katty | oh my |
| 18:38.58 | carrar | yeah |
| 18:39.06 | carrar | no one really owns their land in the USA |
| 18:39.11 | Katty | nods sadly |
| 18:39.32 | Katty | citrusfizz: you could probably do a dialstatus check or something |
| 18:39.41 | Katty | citrusfizz: and then dump variables into a bash() |
| 18:42.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.246) |
| 18:43.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:52.40 | Alex_Bkash | Katty: i need someone who has expertise for implementing a asterisk based calling card and pinless service |
| 18:52.58 | Katty | you might try #asterisk-consultants |
| 18:53.01 | Katty | i think theyr'e still around |
| 18:53.36 | Alex_Bkash | Hello Penguin: |
| 18:53.49 | Alex_Bkash | you are in consultants too |
| 18:53.53 | Alex_Bkash | :) |
| 18:54.51 | Penguin | You've indicated that you don't want to authenticate your callers. I think that's a bad idea. |
| 18:55.05 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.246) |
| 18:55.15 | WIMPy | Or just buy one of those Images with Asterisk and some billing stuff. |
| 18:55.17 | Alex_Bkash | ok |
| 18:55.24 | WIMPy | One click to get rich. |
| 18:55.29 | WIMPy | Maybe. |
| 18:55.29 | Alex_Bkash | :) |
| 18:55.48 | Alex_Bkash | there are many pinless solutions |
| 18:56.01 | Alex_Bkash | they authinticate using CLI |
| 18:56.22 | *** join/#asterisk SGjunior (~sgjunior@out-pq-233.wireless.telus.com) |
| 18:56.25 | carrar | because that is totally safe |
| 18:56.38 | WIMPy | Super safe. |
| 18:56.58 | *** join/#asterisk transfinite (~aleph_0@unaffiliated/transfinite) |
| 18:56.59 | Alex_Bkash | we will use DID |
| 18:57.24 | WIMPy | What does that mean in this context? |
| 18:57.28 | Penguin | That's just the phone number that people dial from their phone. I could acidentally dial it and make calls on your system. |
| 18:58.10 | Alex_Bkash | phn number will be entered into system |
| 18:58.36 | Alex_Bkash | Actually Penguin, WIMPy i dont have clear idea |
| 18:58.42 | Penguin | I can see that. |
| 18:58.46 | carrar | heh |
| 18:58.51 | Alex_Bkash | thats y i need help from some one |
| 18:58.57 | Alex_Bkash | paid support |
| 18:59.04 | Penguin | I thought you either have no idea what you want to do... or you are really bad at describing your idea. |
| 18:59.35 | Alex_Bkash | no idea actually |
| 18:59.45 | *** join/#asterisk BinaryMaster (~TheMental@rrcs-97-76-29-246.se.biz.rr.com) |
| 18:59.53 | WIMPy | The normal german business model. |
| 18:59.55 | Penguin | I guess that's alright. At least you admit it. |
| 19:00.17 | BinaryMaster | can anyone point me in the right direction for this problem /etc/init.d/asterisk start |
| 19:00.17 | BinaryMaster | <BinaryMaster> Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) when I tail /var/log/asterisk/full I get WARNING[3233] db.c: Couldn't create astdb table: disk I/O error |
| 19:00.24 | WIMPy | "I heard someone got rich by doing so-and-so. Where do I have to click to get rich the same way?" |
| 19:00.38 | Alex_Bkash | "Pinless" term i hard a couple of days ago :( |
| 19:00.51 | Penguin | PINless |
| 19:00.59 | Penguin | Not having to enter a PIN. |
| 19:01.01 | *** join/#asterisk Quest (~syncsys@pool2-80-213.brain.net.pk) |
| 19:01.02 | Quest | hi |
| 19:01.19 | Alex_Bkash | WIMPy i am doing wholesale business right now |
| 19:01.23 | Quest | how can I know which the ip and mac of a caller by its excention? |
| 19:01.25 | Penguin | I prefer a PIN, though. |
| 19:01.27 | WIMPy | The days when that was a good idea are over. |
| 19:01.55 | Alex_Bkash | may be |
| 19:01.55 | Quest | I recieved a call from 100 extension (I never setup 100 ) how can I know which the ip and mac of a caller by its excention? |
| 19:02.37 | Penguin | You do not receive calls FROM extensions. |
| 19:02.55 | WIMPy | Quest: If you didn't have some Verbose() with some CHANNEL() variables, you dont have the data. |
| 19:02.56 | Quest | yes |
| 19:03.07 | WIMPy | And that |
| 19:03.10 | Penguin | If the SIP address is 100@some.addr, that's your clue. |
| 19:03.15 | Quest | WIMPy, what do you mean |
| 19:03.32 | Quest | Penguin, well how to find out the sip address? |
| 19:03.33 | *** join/#asterisk nam3l3zz (~quassel@78-60-118-31.static.zebra.lt) |
| 19:03.50 | nam3l3zz | bonjour |
| 19:03.53 | Penguin | If your caller id didn't show it, look on the console. |
| 19:03.55 | WIMPy | If you don;t log it, you don't have it. |
| 19:03.59 | Quest | Penguin, i mean where are the logs that tell me that what was the ip / mac of that call recieved from 100 ext |
| 19:04.07 | Penguin | Verbose() and CHANNEL(), like wimpy mentioned. |
| 19:04.17 | WIMPy | By default that's not logged anywhere. |
| 19:04.21 | nam3l3zz | is there a dialplan command to reject a sip invite ? |
| 19:04.29 | WIMPy | You have to do that in your dialplan. |
| 19:04.32 | Quest | am. Verbose () CHANNEL () ? whats that |
| 19:04.36 | WIMPy | nam3l3zz: Hangup() |
| 19:04.44 | Penguin | Verbose() application; CHANNEL() function |
| 19:05.00 | Quest | Penguin, WIMPy so the call cant be traced? |
| 19:05.03 | WIMPy | Quest: 'core show application verbose', 'core show function CHANNEL' |
| 19:05.27 | WIMPy | Quest: Not unless you implemented somethign to do so. |
| 19:05.30 | Quest | WIMPy, hm. so it was not enabled by default? so i cant trace the call? |
| 19:05.37 | Quest | ok. |
| 19:06.23 | Quest | WIMPy, how to use those commands to enable them |
| 19:06.41 | Quest | and then where the logs can be found? |
| 19:06.53 | WIMPy | Quest: for basic configuration, try the |
| 19:06.58 | WIMPy | ~book |
| 19:06.58 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:07.12 | Quest | thanks |
| 19:07.13 | WIMPy | And if you want to written to a log, use Log instead of Verbose. |
| 19:07.32 | Quest | ok |
| 19:07.41 | Quest | WIMPy, and where the log file would be? |
| 19:07.58 | WIMPy | That's what you configure in logger.conf. |
| 19:07.59 | BinaryMaster | Is there any way to track down what could be causing a problem with creating the astdb? |
| 19:08.10 | Quest | WIMPy, ok |
| 19:12.50 | nam3l3zz | WIMPy: worked, thanx |
| 19:18.53 | *** join/#asterisk nmabry (~nmabry@c-50-134-194-241.hsd1.co.comcast.net) |
| 19:24.16 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:24.46 | nmabry | Howdy! I've configured asterisk to generate CEL events for a database backend, but would love to receive those entries over AMI as well. In cel.conf, I set enabled=yes in the [manager] section and restarted asterisk. I expected to start receiving events over AMI with the type "CEL", but I haven't seen any. Any thoughts on what to check next? |
| 19:25.02 | igcewieling | there is an option to send cel to manager |
| 19:25.10 | igcewieling | check the .conf.sample config files. |
| 19:25.51 | WIMPy | Does your AMI user have the read permission for CEL events? |
| 19:26.10 | igcewieling | have you defined any events to watch? |
| 19:26.29 | nmabry | WIMPy: Ahh, probably not (?). Is the name of the permission group "cel"? |
| 19:27.30 | WIMPy | Should be documented in the sample configs. |
| 19:27.34 | WIMPy | (I hope) |
| 19:28.07 | nmabry | igcewieling: I'm currently receiving CEL generated entries in my database as expected. Do I need to configure AMI to subscribe to CEL events specifically? |
| 19:28.39 | igcewieling | nmabry: doesn't look like it. |
| 19:28.53 | igcewieling | how are you looking for CEL via Manager? |
| 19:29.17 | nmabry | WIMPy: Thanks. I do see a permission class for CDR, but not CEL as far as I can tell |
| 19:29.20 | igcewieling | I assume something like PHPAGI or similar. |
| 19:29.37 | WIMPy | urgs |
| 19:29.48 | igcewieling | To test I recommend manually telnettting to the AMI port, logging in and seeing what events you see. |
| 19:29.48 | Nugget | telnet is eeeeeeevil! |
| 19:29.54 | nmabry | lol |
| 19:30.03 | igcewieling | that is an auto script. |
| 19:30.13 | Penguin | ... or is it? |
| 19:30.51 | nmabry | igcewieling: Thanks. I'll give that a shot and report back. It's possible that our AMI monitor app is misconfigured. |
| 19:33.34 | igcewieling | telent |
| 19:33.50 | igcewieling | Penguin: it is. Has been for at least 8 years |
| 19:33.59 | Penguin | I know. I've been here. |
| 19:35.05 | jmetro | Telnet is good, i have all my boxes open to telnet. |
| 19:35.32 | Penguin | Too bad you don't have telnetd listening. |
| 19:39.19 | blehxor | I've got two separately originated calls that eventually get bridged and attempt to sync so I can capture specific segments of the call. I'm having alot of trouble with timing around the sync. One approach I took was to have cal-1 playback a signal, and have call-2 backgrounddetect. I've noticed that often if call-2 fails to start its background-detect before call-1 starts playback, call-2 will block (~15 sec) until call-1 finishes pla |
| 19:39.20 | blehxor | yback before it starts its background detect. I expected these to be asynchronous - is that not true? |
| 19:39.21 | jmetro | oh that might help , normally i just plug my mouse & keyboard into the serial port and the monitor turns on |
| 19:40.43 | igcewieling | sync? |
| 19:40.55 | igcewieling | I assume english is not your primary language. |
| 19:41.08 | blehxor | synchronize |
| 19:41.23 | igcewieling | the concept does not apply to Asterisk |
| 19:41.35 | igcewieling | you bridge the call, then you are done. |
| 19:42.27 | blehxor | i've got two separately originated calls, whom hit the same conference bridge |
| 19:42.37 | blehxor | asterisk is not my bridge, its the agent automating calls |
| 19:43.19 | igcewieling | I still don't see where "sync" comes in. |
| 19:44.08 | igcewieling | Why, exactly, are you not using Asterisk's native call leg bridging features? |
| 19:46.06 | igcewieling | in any case, put CLI output of the failed call on a pastebin and hope someone wants to go through all the work to understand the complex system you have built. |
| 19:47.17 | blehxor | Sorry, anyways. I want to test the audio quality of the external bridge. |
| 19:52.22 | blehxor | my two separately originated calls run through a dialplan on different extensions |
| 19:53.23 | blehxor | which dial out and bridge a call to this external bridge |
| 19:54.07 | blehxor | then each call can playback and listen to the other call. I play a signal, the other call hears it. Then I start recording segments for audio quality testing. |
| 19:56.00 | *** join/#asterisk SGjunior (~sgjunior@out-pq-233.wireless.telus.com) |
| 19:56.20 | blehxor | but my issue is when the two calls are bridged and are running commands from their dialplan, they seem to block while the other runs a set of commands, which is throwing my record timing off. |
| 19:56.39 | igcewieling | (3:46:06 PM) igcewieling: in any case, put CLI output of the failed call on a pastebin and hope someone wants to go through all the work to understand the complex system you have built. |
| 19:57.01 | igcewieling | stop saying "bridged" The calls are not bridged, they are both connected to a conference room. |
| 19:57.04 | Penguin | It seems that from my experience, blocking happens per channel, not per asterisk. |
| 19:57.11 | igcewieling | Penguin: it does. |
| 19:57.19 | Penguin | A conference... bridge, perhaps? |
| 19:57.25 | blehxor | lol |
| 19:57.36 | blehxor | ok blockign happens per channel |
| 19:57.37 | igcewieling | Penguin: then say a conference bridge. bridging is something asterisk does with two channels. |
| 19:58.09 | blehxor | I typed alot, trying to conserve |
| 19:58.10 | igcewieling | if you don't say the correct words, you won't get a correct answer. |
| 19:58.28 | blehxor | so if blocking happens per channel |
| 19:59.11 | blehxor | and two calls get thrown in the same channel (each running a dialplan) you expect blocking |
| 19:59.38 | blehxor | thats all I want(ed) to know |
| 19:59.44 | igcewieling | but you are not putting two calls in the same channel |
| 19:59.57 | igcewieling | you are taking two calls and sending them to an external service |
| 20:00.28 | igcewieling | in any case, it is not possible for two calls to be on the same channel. |
| 20:00.39 | blehxor | with Dial, which from what I've read bridges the external call to the same channel? |
| 20:00.54 | igcewieling | no. It bridges the incoming channel with the outgoing channel. |
| 20:01.05 | Penguin | Two channels, bridged together, does not make one channel. |
| 20:01.09 | igcewieling | there are two channels as youd see in "core show channels" |
| 20:01.13 | Penguin | It makes one call, but not one channel. |
| 20:01.13 | blehxor | ah ok |
| 20:01.28 | igcewieling | almost all calls have two legs (channels). |
| 20:02.17 | igcewieling | you even see the bridging message in the CLI with the names of the two legs |
| 20:02.33 | SGjunior | ^ |
| 20:03.24 | nmabry | igcewieling: WIMPy: thanks for the help! telnetting in revealed that we *are* receiving CEL events with the "call" privilege class, so its just our monitoring app that's broken. |
| 20:03.54 | igcewieling | nmabry: that is what I suspected. CEL is new, apps don't support it yet. |
| 20:04.24 | igcewieling | blehxor: have you read the Asterisk Book? |
| 20:04.27 | igcewieling | ~book |
| 20:04.27 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:04.40 | blehxor | Thanks, I understand conceptually now. |
| 20:06.20 | BinaryMaster | Okay so I deleted the astdb.sqlLight3 file and let asterisk rebuild it, now asterisk will start however the phones don't ring and the caller is sent directly to voicemail |
| 20:06.32 | BinaryMaster | Can anyone help? |
| 20:06.37 | igcewieling | "sip show peers" |
| 20:07.00 | igcewieling | and the CLI output |
| 20:07.07 | igcewieling | on a pastebin |
| 20:07.18 | BinaryMaster | okay one sec |
| 20:08.45 | BinaryMaster | http://pastebin.com/PKBubHth |
| 20:09.33 | [TK]D-Fender | BinaryMaster: if you deleted the file, that is where * holds REGISTRTATION info so your phones are all dropped for their "contact" field to return |
| 20:09.39 | [TK]D-Fender | you'll have to re-register your phones |
| 20:09.40 | igcewieling | there is the peers, now we need the cli output of a failed call. If I see FreePBX dialplan I'm putting you on /ignore. |
| 20:09.46 | [TK]D-Fender | there is no point |
| 20:09.51 | [TK]D-Fender | restart the phones or await timeout |
| 20:09.58 | igcewieling | [TK]D-Fender: the phones show registered |
| 20:10.19 | igcewieling | except for 104 of course |
| 20:10.44 | BinaryMaster | okay restarting the phones |
| 20:10.55 | [TK]D-Fender | ok, heading home, long day.... BBIAB |
| 20:11.08 | BinaryMaster | how do I show the cli output of a failed call? |
| 20:12.45 | tm1000 | igcewieling: so much hate. rawr |
| 20:23.36 | Penguin | It doesn't matter if you show the failed call or not. We already know why they failed -- you deleted the registration info. |
| 20:23.38 | *** join/#asterisk citrusfizz (~chatzilla@70.184.40.66) |
| 20:23.59 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 20:26.20 | BinaryMaster | rebooting the phones doesn't seem to fix it still goes straight to voicemail |
| 20:26.36 | Penguin | Did you already restart asterisk? |
| 20:27.32 | BinaryMaster | doing that now |
| 20:29.17 | BinaryMaster | still going to voicemail |
| 20:29.32 | Penguin | Are the phones registered again? |
| 20:29.59 | BinaryMaster | by registered what do you mean? |
| 20:30.19 | BinaryMaster | sorry I'm a network guy this is my first time trying to fix a phone system |
| 20:30.27 | Penguin | How many different things can "registered" mean when talking about asterisk? |
| 20:30.41 | Penguin | Oh, so you don't really know any asterisk terminology. |
| 20:30.47 | BinaryMaster | right |
| 20:31.11 | Penguin | Registration is when a device, in this case it is your phone, sends its contact information to asterisk so that asterisk knows how to reach it. |
| 20:31.44 | Penguin | sip show peers will show if the devices are registered. |
| 20:31.45 | BinaryMaster | okay thats logical, is there a way to find out in * if that is done? |
| 20:31.49 | BinaryMaster | okay |
| 20:32.44 | BinaryMaster | looks the same as the pastebin |
| 20:33.27 | Penguin | By the way, those are TERRIBLE names for your devices. |
| 20:33.51 | Penguin | But that indicates that three of four devices are registered and reachable. |
| 20:33.53 | BinaryMaster | lol this is not my baby I'm just trying to pick up the mess |
| 20:34.28 | *** join/#asterisk mokmeister (~mokmeiste@86-41-106-5-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 20:34.35 | BinaryMaster | okay so is the 104 extension making it so it will not ring? |
| 20:34.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:34.50 | Penguin | This pastebin has nothing to do with extensions. |
| 20:35.11 | Penguin | The only thing it says is that the devices are or are not registered. |
| 20:35.51 | BinaryMaster | so is device 104 making it so it doesn't ring? |
| 20:36.08 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 20:36.14 | Katty | srs answers only is urgent thx |
| 20:36.27 | Penguin | The SIP device with the name of 104 isn't reachable. If you are calling to an extension that dials SIP/104, that would cause it to not work. |
| 20:36.40 | jmetro | katty: did you hear about my printer/paper queue for call agents? |
| 20:36.55 | Katty | jmetro: i did not |
| 20:36.58 | Katty | jmetro: what is it |
| 20:37.24 | BinaryMaster | okay so if number rings to all devices and device 104 is not reachable would that cause it to drop to voicemail? |
| 20:37.29 | jmetro | i wanted to setup a printer queue for asterisk so that incoming calls to my queue would be printed out and delivered to my agents who would then scan them in to their PC to pick up the call. |
| 20:37.36 | Penguin | That would depend on the dial plan. |
| 20:38.14 | Penguin | Let's go back to the request of the failed call. Turn verbose to at least 3, then make the call that ends at voicemail. |
| 20:38.28 | Penguin | Pastebin the verbose output. |
| 20:38.42 | BinaryMaster | which log file would it dump to? |
| 20:38.47 | Penguin | None. |
| 20:38.51 | Katty | jmetro: that.... |
| 20:38.55 | Katty | jmetro: needs clarification |
| 20:39.00 | Penguin | CLI verbose output. |
| 20:39.01 | Katty | jmetro: goes to a print spool folder. |
| 20:39.04 | Penguin | Not a log file. |
| 20:39.08 | Katty | jmetro: printer spits them out? or is it emailed? |
| 20:39.08 | BinaryMaster | okay so is there an interactive mode I can use to do this |
| 20:39.24 | Katty | jmetro: hand-delivered to an agent? |
| 20:39.36 | Penguin | That's what the CLI is. |
| 20:39.37 | jmetro | Katty: the papers will be fed via gravity into a paper plane folding machine |
| 20:39.50 | jmetro | Katty: i was originally going to vacuum-tube the paper to the agents but igcewieling found the paper plane machine |
| 20:40.01 | Katty | YES |
| 20:40.05 | Katty | i love this delivery method. go on |
| 20:40.06 | BinaryMaster | sorry I have a freepbx gui so if I ssh into the box how do I enter CLI |
| 20:40.17 | Penguin | asterisk -r |
| 20:40.25 | Katty | jmetro: can we also eject the cdrom drive of the user everytime a call comes in? ALL THE USERS |
| 20:40.28 | [TK]D-Fender | jmetro: no, that was me that linked you the video :) |
| 20:40.39 | jmetro | Katty: Oh it was d-fender =p |
| 20:41.01 | jmetro | Katty: Honestly this method of call queues might be preferable to using app_queue |
| 20:41.07 | Katty | i agree. |
| 20:41.15 | Katty | especially if it will plink them in the back of the head |
| 20:41.20 | Penguin | Please recycle. |
| 20:41.21 | jmetro | Only the tall ones. |
| 20:41.33 | BinaryMaster | holy cow thats a lot of info |
| 20:41.35 | BinaryMaster | lol |
| 20:41.42 | Katty | now if you can also attach a barcode sheet |
| 20:41.52 | Katty | you could setup a scanner somewhere, that reads the barcode location |
| 20:42.00 | Katty | and then setup a delivery route |
| 20:42.14 | Katty | something tells me you could do that with a pi |
| 20:42.37 | jmetro | http://pastebin.com/picEYybA |
| 20:42.44 | Katty | idly wonders if you could could get the delivery route setup with a roomba, and vacuum the floor at the same time |
| 20:42.55 | jmetro | I was thinking the vacuum tube delivery, like at banks. |
| 20:43.46 | Katty | teehee |
| 20:43.48 | BinaryMaster | http://pastebin.com/46EGuJf6 |
| 20:43.52 | Katty | i love it! |
| 20:44.04 | Katty | but roomba delivery service would also be nice |
| 20:44.37 | jmetro | Katty: Oh, you got your voicemail thingy redirecter working |
| 20:44.46 | [TK]D-Fender | Katty: Quad-copter drone is where it's at now.... |
| 20:44.56 | Katty | jmetro: yes, forever ago |
| 20:45.28 | Katty | jmetro: cause i can read documentation! |
| 20:45.30 | Katty | jmetro: go me! |
| 20:46.08 | Katty | wonders what other IP based gadgetry support open ssh keys |
| 20:46.26 | Katty | oh things you could do with asterisk and a pi |
| 20:46.30 | Penguin | binarymaster: It says the extensions are not available. I have to assume this is because the SIP devices are unreachable. |
| 20:46.44 | BinaryMaster | I can ping them |
| 20:46.56 | Katty | yay they respond to icmp! |
| 20:46.58 | jmetro | Katty: Setup a showerhead with a Pi that reads temperature, turn the shower on, receive a text message when it reaches the correct temperature. |
| 20:46.59 | Katty | that's a good first start |
| 20:47.00 | BinaryMaster | and I can log into their web interfaces |
| 20:47.07 | [TK]D-Fender | BinaryMaster: dialparties.agi: Setting default NOANSWER DIALSTATUS since no extensions availa ble |
| 20:47.14 | Katty | jmetro: i can see it now |
| 20:47.22 | [TK]D-Fender | BinaryMaster: you just blew away AstDB, didn't you? |
| 20:47.22 | Katty | jmetro: root@foo.bar HOT HOT HOT |
| 20:47.28 | jmetro | XD |
| 20:47.33 | BinaryMaster | yes |
| 20:47.40 | jmetro | Except it'd be more like |
| 20:47.43 | jmetro | Shower@soap.bar |
| 20:47.44 | [TK]D-Fender | BinaryMaster: that SCREWS the values FreepBX puts in there |
| 20:47.51 | *** join/#asterisk TimeRider (~steve@90.213.18.65) |
| 20:47.52 | [TK]D-Fender | BinaryMaster: It doesn't even WANT to try |
| 20:48.04 | [TK]D-Fender | there is a script te regenerate it you have to run |
| 20:48.05 | BinaryMaster | Is there a way to force it to rebuild it |
| 20:48.07 | Penguin | And this is why we don't use FreePBX. Too much bother. |
| 20:48.14 | BinaryMaster | :( |
| 20:48.25 | BinaryMaster | I hate getting other peoples crap :( |
| 20:48.44 | Katty | jmetro: toaster@home.bar YOUR TOAST IS READY MA"AM |
| 20:48.55 | BinaryMaster | Okay so can you point me to the script or should I ask the FreePBX guys |
| 20:49.06 | Katty | jmetro: frontdoor@home.bar OMG DISARMED!!!! |
| 20:49.09 | tm1000 | BinaryMaster: it's not crap |
| 20:49.23 | BinaryMaster | I'm sure it isn't |
| 20:49.23 | tm1000 | you are just in a purist channel. not sure why you'd expect everyone to help |
| 20:49.26 | [TK]D-Fender | http://www.minezone.nl/node/5 |
| 20:49.37 | tm1000 | the only person who is helping is TK and he's in #freepbx anyways |
| 20:49.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:49.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:50.01 | BinaryMaster | lol thank you |
| 20:50.12 | jmetro | Katty: at this point we turn into Human-jabba the hutts. |
| 20:50.23 | tm1000 | point is keep freepbx things in #freepbx and you wont be attacked or bagered by purists |
| 20:50.44 | [TK]D-Fender | or worse ... RABID WEASELS |
| 20:54.25 | Katty | like mister fender here. |
| 20:54.31 | Katty | he's a very weasely sort. |
| 20:55.09 | jmetro | not an icy one at least. |
| 20:55.13 | jmetro | those are the worst. |
| 20:55.44 | BinaryMaster | okay so now it doesn't go to voice mail any more ... but it still doesn't ring now the call just gets dropped |
| 20:56.01 | jmetro | watch the CLI find out where it goes wrong |
| 20:56.51 | BinaryMaster | Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/102-00000005' in macro 'hangupcall' |
| 20:56.51 | BinaryMaster | <PROTECTED> |
| 20:56.51 | BinaryMaster | <PROTECTED> |
| 20:56.51 | BinaryMaster | <PROTECTED> |
| 20:56.55 | tm1000 | OH NO |
| 20:56.56 | jmetro | Nooooo |
| 20:57.00 | tm1000 | lol |
| 20:57.06 | tm1000 | dont paste to a channel |
| 20:57.16 | BinaryMaster | so what exactly does that part mean? |
| 20:57.20 | jmetro | core set verbose 10 |
| 20:57.22 | jmetro | core set debug 10 |
| 20:57.25 | tm1000 | lol |
| 20:57.26 | jmetro | copy the WHOLE THING |
| 20:57.29 | jmetro | and put it in a pastebin link |
| 20:57.30 | BinaryMaster | okay |
| 20:57.31 | tm1000 | DONT POST THOUGH |
| 20:57.35 | jmetro | PASTE THE BIN OUT OF IT |
| 20:57.42 | tm1000 | ITS FUN TO CAPS |
| 20:58.04 | jmetro | Even with cruise control you still have to steer. |
| 20:58.23 | Katty | i'll steer your cruise control in a minute. |
| 20:58.38 | tonyclewis | Penguin: who do you mean by "We" |
| 20:58.49 | tonyclewis | the Asterisk community |
| 20:58.51 | tonyclewis | Developers |
| 20:58.52 | tonyclewis | or you |
| 20:58.55 | Penguin | .esrever ni epyt ot nuf osla s'tI |
| 20:59.00 | [TK]D-Fender | BinaryMaster: First, don't flood junk here... ever. next that part means NOTHING. that is just end-of-call cleanup junk that FreePBX starts processing |
| 20:59.02 | tonyclewis | as that statement is very misleading |
| 20:59.06 | tm1000 | tonyclewis: his company |
| 20:59.17 | tm1000 | err Penguin's company |
| 20:59.23 | [TK]D-Fender | BinaryMaster: It says precisely nothing about what it was doing up until the end |
| 20:59.29 | Penguin | I mean people with sense, in general. |
| 20:59.35 | tonyclewis | ahhh ok |
| 20:59.40 | tonyclewis | just wanted clarification |
| 20:59.49 | tm1000 | yet you hang out in #freepbx…Penguin baffles the mind :-) |
| 20:59.49 | *** join/#asterisk J4nus (~janus@cable-213.214.36.245.coditel.net) |
| 20:59.51 | BinaryMaster | http://pastebin.com/zbWB76XA |
| 21:00.23 | [TK]D-Fender | BinaryMaster: and that is a pile of CORE debug crap you don't need |
| 21:00.38 | [TK]D-Fender | BinaryMaster: You should almost never need more than basic verbose, and frequently SIP debug |
| 21:00.47 | tonyclewis | because last I knew that was not Digiums stance |
| 21:00.57 | BinaryMaster | level 3 then? |
| 21:00.59 | tonyclewis | so wanted to make sure someone did not think you were speaking in behalf of Asterisk |
| 21:01.07 | jmetro | Katty: I'll stick your shift? |
| 21:01.08 | Penguin | binarymaster: core set debug off |
| 21:01.11 | J4nus | Hi, I have a very strange problem. When I use my softphone and 3G connection, I have no sound. I see from the statistic there is no download bandwidth. If I use Wifi it's working. Now If in place of doing a Dial, I use a Playback cmd, it's working |
| 21:01.22 | [TK]D-Fender | BinaryMaster: And that particular little snippet was about an OPTIONS packet and not part of an actual call... and probably says nothing about anything anone should ahve reason to care about |
| 21:01.33 | Penguin | And verbose level 3 is usually enough, but sometimes higher level is needed. |
| 21:01.40 | J4nus | and I have another server with similar config and there it's working.... I spent already 1 day to search the problem but I don't find the root cause |
| 21:02.03 | [TK]D-Fender | J4nus: you probably have a REINVITE issue where your endpoints are trying to comunicate directly and that is the point of failure |
| 21:02.38 | [TK]D-Fender | J4nus: never think "other server is fine". Your NETWORKING config and peer setup are not agreeing with the situation |
| 21:03.16 | J4nus | i'm in directmedia=no |
| 21:03.35 | J4nus | and if it's a reinvite issue, why does it work with Playback ? |
| 21:03.39 | BinaryMaster | http://pastebin.com/mZ6sJB7R |
| 21:04.17 | J4nus | In fact, my softphone is registered in an asterisk, this asterisk is connected to another asterisk which is acting as a media gateway (gateway to pstn world) |
| 21:04.40 | J4nus | and on the media gw, in place of Dial(DAHDI..) I did Playback(hello-world) |
| 21:04.47 | J4nus | and i heard "hello world" |
| 21:04.58 | J4nus | now if I change back to Dial.. No sound.. |
| 21:05.11 | Katty | sometimes i'm really surprised. |
| 21:05.18 | Katty | i mean, i know i take things for granted.... |
| 21:05.20 | [TK]D-Fender | BinaryMaster: they rang. nobody answered |
| 21:05.26 | J4nus | the only difference i see between 3G and Wifi: 3G it's a public ip. Wifi it's a natted ip |
| 21:05.26 | Katty | but sometimes i really wonder about people in here |
| 21:06.14 | [TK]D-Fender | J4nus: the answer is not changing. if over the same tech a DIAL doesn't give audio both ways, and a PLAYBACK does then it is a NETWORKING misconfiguration between * and your endpoints |
| 21:06.34 | [TK]D-Fender | J4nus: You have not configured your *'s right |
| 21:06.40 | jmetro | Katty: i had to explain how to double click to someone. I wonder about nobody :| |
| 21:06.49 | Katty | oh my. |
| 21:07.01 | [TK]D-Fender | jmetro: Or conversely... everybody :) |
| 21:07.26 | J4nus | [TK]D-Fender, so it's a problem of NAT you think ? |
| 21:07.33 | BinaryMaster | if no one picks up it should drop to extension 103's voicemail |
| 21:07.39 | Katty | also, so far my little vmware instance of asterisk is doing ok |
| 21:07.42 | J4nus | I looked the firewall and i don't see any drop |
| 21:07.43 | [TK]D-Fender | J4nus: yes |
| 21:07.54 | [TK]D-Fender | J4nus: Look at the actual; SIP trace. |
| 21:07.55 | Katty | it's not barfing, which is always good. |
| 21:08.03 | [TK]D-Fender | J4nus: Look at your actual general settings |
| 21:08.04 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 21:08.25 | [TK]D-Fender | J4nus: you've also told us nothing about where any of these pieces are relative to each other |
| 21:09.40 | J4nus | the main asterisk server is a public ip, directly routed from internet |
| 21:09.52 | J4nus | i made this choice to avoid natting issues.. |
| 21:10.04 | jmetro | Katty: VM-asterisk is all we run, it works phenomenally (⌐■‿■) |
| 21:10.23 | J4nus | i took wireshark trace but i'm not good enough to investigate into the issue |
| 21:10.48 | [TK]D-Fender | J4nus: forget wireshark and look at ASTERISK SIP debu |
| 21:11.06 | Katty | jmetro: how many boxes are you running vsphere? |
| 21:11.18 | jmetro | Ridiculous amounts, but we have 7 asterisks. |
| 21:11.26 | Katty | jmetro: are they all in the same vlan? |
| 21:11.31 | jmetro | hm |
| 21:11.44 | jmetro | I believe so |
| 21:11.53 | Katty | jmetro: i'm hoping to segregate ours |
| 21:11.58 | Katty | jmetro: going to have 3 when i'm done |
| 21:12.11 | jmetro | Should work fine |
| 21:12.21 | Katty | jmetro: and i'm not entirely sure how to register a box in one vlan as a peer on another vlan |
| 21:12.46 | jmetro | ahhh youre crossing them together |
| 21:12.49 | jmetro | ours are individual |
| 21:13.13 | Katty | well box A is going to have a massive crapton of sip channels from bandwidth or somewhere |
| 21:13.24 | Katty | and boxes B-whatever will be clients |
| 21:13.29 | jmetro | hm |
| 21:13.30 | Katty | tenants, whatever. individual companies |
| 21:13.51 | Katty | and box a becomes the sip providor for the rest |
| 21:14.02 | Katty | i'll figure it out. somehow. |
| 21:14.23 | jmetro | we template it up, one can go down without affecting the others |
| 21:14.28 | Katty | nods |
| 21:15.11 | jmetro | and then backup-boxes of all that |
| 21:15.19 | jmetro | so really 14 boxes |
| 21:15.52 | Katty | fancy. |
| 21:19.25 | Katty | i'm debating whether or not to do it with a VPN tunnel. |
| 21:19.37 | Katty | and i think that's probably the easiest way, at least with what we're doing |
| 21:20.26 | jmetro | ^ |
| 21:20.27 | jmetro | yes |
| 21:21.07 | [TK]D-Fender | [17:07]BinaryMasterif no one picks up it should drop to extension 103's voicemail <- then you have misconfigured the ring-group |
| 21:26.17 | *** part/#asterisk AL13N (~alien@2001:770:16f:0:225:22ff:fe40:2e46) |
| 21:30.55 | jmetro | If i dial a local channel that executes a macro when the party picks up and then also attach a macro-on-pickup to the channel, what happens |
| 21:31.07 | jmetro | essentially, Dial(local/announcer,M(announce2) |
| 21:31.13 | jmetro | and announcer is |
| 21:31.14 | Penguin | The asterisk caves in on itself. |
| 21:31.23 | jmetro | dial(sip/person,M(announce1)) |
| 21:32.46 | [TK]D-Fender | jmetro: Close..... |
| 21:33.39 | dpeloquin | so anyone going to astricon? |
| 21:33.43 | jmetro | [TK]D-Fender: close to what? I dont know what happens. |
| 21:34.01 | [TK]D-Fender | jmetro: Helps when you remember your syntax |
| 21:34.56 | jmetro | well |
| 21:35.01 | jmetro | thats pseudocode for one. |
| 21:35.23 | jmetro | im just asking what happens when you dial a local channel and both channels have announcements on pickup |
| 21:40.19 | [TK]D-Fender | they'll stack |
| 21:40.38 | [TK]D-Fender | the internal should get called first with M() |
| 21:40.46 | [TK]D-Fender | then the outer dial. |
| 21:41.02 | [TK]D-Fender | as it should only go on one the macro exits and then bridges |
| 21:42.09 | jmetro | Ugh |
| 21:42.31 | jmetro | thats annoying. Trying to ring two cells and 4 desk phones with differeing announcements. |
| 21:43.28 | [TK]D-Fender | time to do it all in nested levels |
| 21:44.07 | jmetro | huh |
| 21:45.38 | BinaryMaster | Thank you everyone that helped |
| 21:53.30 | *** join/#asterisk SGjunior (~sgjunior@out-pq-233.wireless.telus.com) |
| 21:58.01 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 22:10.12 | *** join/#asterisk tzafrir (~tzafrir@2001:41b8:203:dc13:6267:20ff:fe51:8328) |
| 22:23.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.225) |
| 22:27.10 | jmetro | i love that linksys-cold-restart works perfectly on cisco phones but the cisco reboot does 0 things. |
| 22:34.03 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 22:36.15 | *** join/#asterisk SGjunior (~sgjunior@out-pq-233.wireless.telus.com) |
| 22:37.58 | *** join/#asterisk level7 (~chatzilla@31.44.17.249) |
| 22:38.06 | level7 | hi all.....anybody using sipwise? |
| 22:39.10 | *** join/#asterisk psanders (~psanders@ks-184-2-115-125.dhcp.embarqhsd.net) |
| 22:46.41 | newtonr | level7, haven't heard anyone mention it. Is it an ITSP? |
| 22:47.04 | newtonr | looks it up |
| 22:47.59 | level7 | http://sipwise.com/ |
| 22:48.50 | newtonr | yeah I see. Interesting. Nope, I have no experience with it. |
| 22:49.45 | level7 | there is a "test" appliance for virtualbox |
| 22:49.51 | level7 | I am testing it, it looks nice |
| 22:59.33 | *** join/#asterisk mitchrodrigues (~mitchrodr@38.111.144.81) |
| 23:02.40 | jmetro | i should /nick level3 |
| 23:09.17 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 23:10.55 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:15.31 | Katty | WOCKA. |
| 23:15.33 | Katty | that is all. |
| 23:17.42 | cusco | http://rumpetroll.com/ funny weird stuff |
| 23:25.13 | Katty | what do i want to make for dessert? lemon meringue, tiramisu, rum cake, or choc chip cookies |
| 23:25.32 | *** join/#asterisk SGjunior (~sgjunior@out-pq-233.wireless.telus.com) |
| 23:29.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.247) |
| 23:33.35 | cusco | err |
| 23:33.48 | cusco | tiramissu would go easy now |
| 23:33.49 | cusco | :) |
| 23:37.52 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 23:39.41 | Penguin | If you're asking what I want for dessert, I'd like some peanut butter cookies. |
| 23:40.11 | cusco | lol |
| 23:41.09 | WIMPy | has some halva with almond |
| 23:43.12 | carrar | Katty, tiramisu |
| 23:43.16 | carrar | IF YOU DO IT RIGHT |
| 23:44.09 | carrar | Or Godiva Cheese Cake |
| 23:44.10 | Penguin | With homemade lady fingers? |
| 23:44.35 | carrar | well Feed to me with lady fingers |
| 23:45.00 | carrar | lady fingers attached to a lady |
| 23:45.20 | carrar | Just Clarifying |
| 23:45.29 | Katty | i know just the lady. |
| 23:45.30 | carrar | heh |
| 23:46.41 | carrar | Katty |
| 23:46.42 | carrar | http://www.tasteofhome.com/recipes/cream-cheese-chocolate-cupcakes |
| 23:46.46 | carrar | I just ate one of those |
| 23:46.55 | carrar | My mom made me some |
| 23:47.01 | carrar | they are so good |
| 23:47.03 | Katty | taste of home is always good. haven't had a bad one yet |
| 23:47.04 | Katty | looks |
| 23:47.31 | Katty | yesh, those look tasty!! nomnom |
| 23:48.17 | carrar | not loosing any weight eating those |
| 23:48.42 | carrar | but I walked 4.5 miles today so I don't feel so bad |
| 23:48.47 | Katty | yay! |
| 23:48.51 | Katty | i'm about to workout here in a minute |
| 23:49.34 | Katty | it's weight day! |
| 23:50.07 | carrar | OH NO |
| 23:50.11 | carrar | Not that day |
| 23:50.16 | Katty | i don't mind weight day so much |
| 23:50.23 | carrar | Are you the biggest WINNER |
| 23:50.28 | carrar | WINNING |
| 23:50.31 | Katty | well. |
| 23:50.36 | Katty | it's not really a competition |
| 23:50.40 | carrar | oh |
| 23:50.45 | carrar | it's always a competition |
| 23:50.50 | Katty | i'm just gonna sit here and do my thing |
| 23:50.56 | Katty | and then i get an insta-win? |
| 23:51.04 | carrar | no gold stars? |
| 23:51.19 | Katty | i suppose i could go cut some out of construction paper and tape it to myself. |
| 23:51.26 | Katty | that might work |
| 23:51.28 | carrar | You totally should |
| 23:51.37 | Katty | i will do that when i hit my goal then |
| 23:51.41 | Katty | and put it on fb |
| 23:51.42 | Katty | just for you! |
| 23:51.46 | carrar | yeah |
| 23:51.48 | carrar | fb |
| 23:51.52 | carrar | since I am not on fb |
| 23:51.56 | Katty | :< |
| 23:52.05 | carrar | I have my own web site |
| 23:52.08 | carrar | no need for fb |
| 23:52.08 | Katty | well i'll link it to you |
| 23:52.13 | carrar | heh |
| 23:53.29 | *** join/#asterisk SGjunior (~sgjunior@out-pq-198.wireless.telus.com) |