| 02:31.24 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 02:31.24 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.11 (2009/11/30), 1.6.0.19 (2009/11/30), 1.4.27.1 (2009/11/30), *-Addons 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 02:31.37 | bcrisp | ha i checked using an online port tester |
| 02:32.21 | MarkJenks | Is there a bugtracker for asterisk-gui where I can post some problems I have found? |
| 02:33.03 | MarkJenks | I put them on the list, but is seems that it isn't followed anymore |
| 02:33.16 | bcrisp | udp appears open from the server but using a port tool it says its closed |
| 02:33.28 | bcrisp | the firewall has an exception allowing it be open |
| 02:34.00 | MarkJenks | what does netstat say? |
| 02:34.09 | MarkJenks | is it bound? |
| 02:34.11 | bcrisp | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 02:34.27 | MarkJenks | bound to any, that's good |
| 02:34.40 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 02:34.51 | MarkJenks | are you running iptables? |
| 02:34.58 | bcrisp | i dont know :/ linux newb here |
| 02:35.06 | MarkJenks | iptables --list |
| 02:35.43 | bcrisp | ok let me pastebin |
| 02:36.18 | bcrisp | http://pastebin.ca/1707801 |
| 02:36.22 | bcrisp | it appears so |
| 02:37.48 | bcrisp | reject all anywhere anywhere looks suspicious.. |
| 02:38.01 | MarkJenks | I believe the any any at the top opens all, but I wonder about the specify udp |
| 02:38.18 | bcrisp | the very bottom line says reject all from anywhere right? |
| 02:38.21 | bcrisp | (to anywhere) |
| 02:38.37 | MarkJenks | yeah, that's normal. All firewall should have a reject all at the end |
| 02:38.45 | bcrisp | i dont want a software firewall |
| 02:38.53 | bcrisp | i have a hardware firewall.. can i disable this? |
| 02:38.54 | MarkJenks | you on fedora 12? |
| 02:38.57 | bcrisp | centos 5 |
| 02:39.00 | kam187 | just turn it off |
| 02:39.09 | bcrisp | iptables -- ofF? |
| 02:39.23 | kam187 | service iptables off |
| 02:39.26 | kam187 | for now |
| 02:39.31 | kam187 | chkconfig iptables off |
| 02:39.34 | kam187 | forever |
| 02:39.48 | MarkJenks | yep |
| 02:40.01 | bcrisp | working now :) |
| 02:40.08 | MarkJenks | kam187 is faster than me. ;) |
| 02:40.09 | bcrisp | im seeing the flood of debug messages for sip |
| 02:40.19 | kam187 | [02:18] <kam187> u sure iptables isnt running or something now? |
| 02:40.27 | kam187 | [02:39] <bcrisp> working now :) |
| 02:40.28 | kam187 | lol |
| 02:40.48 | bcrisp | the other install didnt have the iptables running |
| 02:40.50 | bcrisp | this one did |
| 02:41.06 | MarkJenks | Alot of newer distros have it on my default. |
| 02:41.15 | MarkJenks | you can choose on/off during the install |
| 02:41.24 | MarkJenks | network config part |
| 02:41.27 | bcrisp | the other one was centos 5 w / plesk , this was just another image |
| 02:41.45 | bcrisp | cisco firewall |
| 02:41.49 | bcrisp | well thanks for helping me with that |
| 02:41.56 | MarkJenks | asa 8.2 rocks |
| 02:42.00 | kam187 | hw fw ftw |
| 02:42.17 | jblack | Found something you guys can can all get your girlfriends... usb breast warmers. http://www.youtube.com/watch?v=50jLMM860T4 |
| 02:42.18 | kam187 | omg i'm a geek |
| 02:42.51 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 02:43.00 | MarkJenks | geek? I'm too old to be a geek |
| 02:43.37 | *** join/#asterisk LemensTS (n=customgt@adsl-99-28-223-95.dsl.stl2mo.sbcglobal.net) |
| 02:43.44 | kam187 | hehe |
| 02:44.08 | MarkJenks | 1983 there were no geeks, just hackers. :) |
| 02:44.37 | kam187 | yup |
| 02:44.57 | TimRiker | [TK]D-Fender, net splits are causing the bot grief. It sometimes fails to reconnect on some of it's nicks. :( |
| 02:44.57 | MarkJenks | OMG, that was 26 years ago. |
| 02:45.47 | kam187 | lol |
| 02:45.58 | kam187 | anyone still use slackware? |
| 02:46.02 | kam187 | havnt used that in ageeeeeeeeeeeeees |
| 02:46.28 | MarkJenks | cpm ftw |
| 02:46.46 | MarkJenks | I haven't looked at slack in years |
| 02:47.08 | MarkJenks | even then, not much |
| 02:47.44 | MarkJenks | hate to bring it up again, but any good resources for bugs in the GUI? |
| 02:48.02 | bcrisp | yay meetme working |
| 02:48.12 | bcrisp | thanks guys, have a good night |
| 02:48.15 | MarkJenks | gn |
| 02:50.04 | nitrus^ | do drop calls after 30 seconds usually indicate some sort of network or NAT problem? |
| 02:52.23 | MarkJenks | Wanna read a good Article about the beginning of IM? http://im.about.com/od/imbasics/a/history-of-im-interview.htm |
| 02:52.43 | MarkJenks | Is it same same 30 all the time? |
| 02:53.01 | MarkJenks | r/same same/the same |
| 02:53.07 | spiegel | kam187 i am |
| 02:53.12 | nitrus^ | yeah |
| 02:53.21 | nitrus^ | 30 seconds everytime pretty much |
| 02:53.32 | *** join/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com) |
| 02:53.51 | MarkJenks | if you grep your extensions.conf for 30, does anything stand out? |
| 02:54.30 | LemensTS | how do you turn sip debug off |
| 02:54.47 | nitrus^ | sip set debug off |
| 02:54.47 | LemensTS | nm |
| 02:55.02 | nitrus^ | no, nothing stands out |
| 02:55.18 | nitrus^ | im running freepbx so it pretty much does everything for me |
| 02:55.25 | nitrus^ | could it be a context issue or something? |
| 02:55.37 | MarkJenks | if you turn your verbose to 6 it might tell ya something more |
| 02:55.51 | nitrus^ | wouldnt i see the same messages at 100? |
| 02:56.05 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 02:56.14 | kam187 | spiegel: how is it? |
| 02:56.28 | LemensTS | nitrus: does it do it from sip phone to sip phone? |
| 02:58.12 | nitrus^ | ahh good question |
| 02:58.15 | nitrus^ | lmmie check |
| 02:58.32 | MarkJenks | yeah, that would seperate a config vs trunk issue |
| 03:12.00 | iconicflux | alright.. I want to know what sick fucker has made an annoying MMS generator and given it to my brother. |
| 03:12.31 | MarkJenks | night all |
| 03:20.23 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 03:25.36 | *** join/#asterisk d00gster (n=doughant@94.98.134.132) |
| 03:31.45 | *** join/#asterisk _bugz_ (n=bugz@adsl-76-194-243-94.dsl.lsan03.sbcglobal.net) |
| 03:33.21 | *** join/#asterisk mintos (n=mvaliyav@117.206.18.98) |
| 04:04.19 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:04.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:13.00 | *** join/#asterisk OmbreNoir (n=OmbreNoi@modemcable070.50-177-173.mc.videotron.ca) |
| 04:17.06 | *** join/#asterisk jtodd (i=csii1j32@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 04:17.06 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 04:31.49 | *** join/#asterisk MarkJenks (i=ce2871c7@gateway/web/freenode/x-ppgewvesrgtrzcuv) |
| 04:33.42 | *** part/#asterisk MarkJenks (i=ce2871c7@gateway/web/freenode/x-ppgewvesrgtrzcuv) |
| 04:33.50 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 04:48.58 | *** join/#asterisk Chotaire (i=chotaire@to.to.to) |
| 04:50.15 | *** join/#asterisk thuddwhirr (i=4b656f82@gateway/web/freenode/x-wyiqffdswxpwwvrf) |
| 04:50.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:51.17 | thuddwhirr | hello! anyone here use asterisk with a sip proxy like openser? |
| 04:52.18 | *** join/#asterisk geneticx (n=geneticx@adsl-144-252-204.mia.bellsouth.net) |
| 04:52.43 | thuddwhirr | i'm having an odd issue. i have outgoingproxy set in my sip.conf. When I make outgoing calls, everything behaves fine, with all SIP routing through the proxy. When i receive an incoming call though, the SIP initially routes through the proxy, everythign from the invite to the 200 OK |
| 04:53.03 | thuddwhirr | but the ACK and BYE bypass the proxy and go directly to the other UA |
| 04:54.47 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 05:01.30 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 05:12.03 | *** join/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) |
| 05:20.48 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:34.14 | *** join/#asterisk simplydrew_ (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 05:35.56 | *** join/#asterisk mintos (n=mvaliyav@117.206.18.129) |
| 05:45.30 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 05:48.00 | *** join/#asterisk Tim_Toady (n=moi@188.4.76.160.dsl.dyn.forthnet.gr) |
| 05:49.33 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 06:01.55 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 06:07.13 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 06:08.30 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 06:27.24 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 06:32.36 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 06:41.53 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 06:44.41 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 06:50.10 | *** join/#asterisk vk2dgy (n=rossw@ali-syd-3.albury.net.au) |
| 06:50.17 | *** join/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 06:50.57 | vk2dgy | very quiet for so many users... lots of lurkers? |
| 06:51.01 | [T]ank | anyone here ever set up a polycom soundstation using just the web interface rather than the tftp server? |
| 06:51.27 | toresbe | vk2dgy: Everyone's on the phone, duh. :) |
| 06:51.32 | vk2dgy | lol. |
| 06:52.10 | *** join/#asterisk ticoit (n=ticoit@201.191.190.238) |
| 06:52.14 | vk2dgy | I came in here to see if anyone might be able to advise me how I can issue more than one request using "asterisk -rx "command1; command2" - or even it it's possible? |
| 06:52.53 | [T]ank | im having a hard time getting my new polycom soundstation ip 6000 to register. it keeps giving me a user / password mismatch. currently waiting for it to come back to life after the 10 minute restart so i can copy the error and share it. |
| 06:53.10 | vk2dgy | spcifically, I want to issue at least two (possibly more) commands: |
| 06:53.13 | vk2dgy | sip show peers |
| 06:53.15 | vk2dgy | show channels |
| 06:53.27 | [T]ank | im missing a field in the configuration but i cannot figure out which one. |
| 06:53.34 | vk2dgy | and two calls to asterisk -rx seems to be a terrible waste of resources. |
| 06:53.47 | toresbe | vk2dgy: that's premature optimization if I've ever seen it |
| 06:54.00 | vk2dgy | s/two/& or more/ |
| 06:54.44 | vk2dgy | yes, this keeps happening periodically. |
| 06:55.29 | vk2dgy | since it's done as a ssh call to a remote server, less ssh calls, less calls to asterisk, can only be a good thing. |
| 06:55.56 | vk2dgy | since the outputs are just contatenated to the same file for subsequent processing anyway.... if I could run (multiple) commands in one hit it'd be much more efficient. |
| 06:56.31 | vk2dgy | I tried seperating commands with & and , and also with \n (and expanded the line to put in a hard return), nothing worked. |
| 06:57.01 | vk2dgy | (comma should have been semicolon (;) above) |
| 06:57.49 | [T]ank | WARNING[10727]: chan_sip.c:8272 check_auth: username mismatch, have <Polycom>, digest has <> |
| 06:58.50 | [T]ank | I know the peer on asterisk is set up correctly because I have a couple of other soundpoint ip phones connecting to them. I set them up using ftp, but the server is going away... i just copied their peer and set this up using the web interface instead. |
| 06:59.01 | [T]ank | what field could i possibly be missing? |
| 06:59.08 | [T]ank | digest is blank. |
| 06:59.15 | [T]ank | what missing field would cause that? |
| 07:07.26 | ChannelZ | [T]ank: it looks like the device is trying to authenticate but with no auth username (this is different than the SIP account it's registering as) |
| 07:07.40 | ChannelZ | why it's blank though I don't know |
| 07:10.15 | [T]ank | its rebooting again... when it comes up i will verify what i have set. |
| 07:12.14 | [T]ank | is there a way to know if this phone is even compatible with asterisk? I would assume it is. |
| 07:13.32 | [T]ank | Auth User ID is populated. It is set to Polycom |
| 07:13.59 | ChannelZ | what does the entry for that phone in sip.conf look like? |
| 07:14.08 | [T]ank | sec... |
| 07:15.33 | [T]ank | http://pastebin.ca/1707997 |
| 07:15.40 | *** join/#asterisk mokmeister (n=mokmeist@109.78.117.86) |
| 07:16.23 | mokmeister | hi all |
| 07:17.11 | [T]ank | ChannelZ: look right? |
| 07:17.42 | mokmeister | we are looking at replacing aging voice mail systems and I was wondering how difficult it would be to implement using asterisk |
| 07:18.00 | mokmeister | We have MD110s as our main pbxs |
| 07:18.29 | mokmeister | we have just upgraded one of our mds with a TSE |
| 07:18.45 | mokmeister | we are going to see how that goes |
| 07:18.48 | ChannelZ | [T]ank: make sure you don't have a space or something typed into the auth username (not the SIP name) on the phone |
| 07:19.06 | mokmeister | And we are currently looking at changing our voice mail system |
| 07:19.48 | mokmeister | Anywhere I get information on this would be greatly appreciated |
| 07:25.14 | *** join/#asterisk MAbbas (i=Jinbaba@115.186.7.5) |
| 07:25.44 | MAbbas | Hi everyone ... |
| 07:26.45 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:27.27 | ChannelZ | [T]ank: Do you have some auth= line in your sip.conf somewhere earlier? post the whole thing |
| 07:29.23 | ChannelZ | [T]ank: or what version of * are you running? The only reason I can think you would be getting this behavior is if you had an auth line in your sip.conf specifically and the phone is not sending an auth, or something changed in 1.6 which is making the username=xxx directive in your sip.conf act like an auth (1.4 doesn't seem to do this so I'm just guessing there) |
| 07:29.56 | *** join/#asterisk xrmx__ (n=rm@host143-252-dynamic.8-87-r.retail.telecomitalia.it) |
| 07:30.12 | MAbbas | I am using AMI in my python code. But the problem is when I send an action, the response does not come in a single tcp packet .. therefore I have no idea how many more response packet I have to read. |
| 07:30.38 | MAbbas | to get full response |
| 07:31.42 | [T]ank | ChannelZ: just deleted everything but that entry and my general settings. |
| 07:31.45 | [T]ank | [general] qualify=yes context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes |
| 07:34.18 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 07:34.34 | ChannelZ | MAbbas: you read until a blank line |
| 07:35.07 | MAbbas | you mean '\r\n'? |
| 07:35.13 | ChannelZ | yes |
| 07:35.41 | ChannelZ | same as how you terminate your commands |
| 07:35.45 | MAbbas | is it true for all commands? |
| 07:35.56 | ChannelZ | it should be |
| 07:35.59 | MAbbas | let me try .. |
| 07:37.03 | ChannelZ | [T]ank: well see additional questions - you could tell for sure by doing a SIP debug and attempting to register the phone, see if it's sending an "Authenticate" header with no user |
| 07:38.59 | vk2dgy | is away: |Auto set-away.| Msgs saved. |
| 07:42.02 | [T]ank | http://pastebin.ca/1708011 |
| 07:43.00 | [T]ank | ChannelZ: what do you think. |
| 07:43.28 | ChannelZ | that shows everything after the fact |
| 07:44.43 | [T]ank | http://pastebin.ca/1708013 |
| 07:45.56 | ChannelZ | ok see line 194 |
| 07:46.23 | ChannelZ | the phone is sending an authorization that is blank |
| 07:47.26 | [T]ank | yeah... thats whats odd though... in the web config i have all of the fields punched in... im stumped. |
| 07:47.56 | ppc | [T]ank: what are you using, freepbx? |
| 07:48.18 | [T]ank | asterisk |
| 07:48.32 | [T]ank | web config for the polycom phone is what I was referring to |
| 07:48.35 | ppc | oh |
| 07:49.07 | ChannelZ | well I don't know anything about that phone, can you post a screenshot of the config page? |
| 07:49.35 | [T]ank | im going to call it a night and try again tomorrow. almost 1am here. need to be back here to work at 8. |
| 07:49.44 | [T]ank | thanks for the help so far |
| 07:50.25 | ChannelZ | Since you're not specifying authentication in sip.conf the authentication username in the phone shouldn't have anything in it (which it seems like it doesn't already if it's not putting it in the Authenticate header, which it shouldn't be sending anyway if it's not on) |
| 07:50.35 | MAbbas | <PROTECTED> |
| 07:52.54 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 07:53.30 | MAbbas | if I send mutiple AMI commands .. (in my python code)how do I map which response belongs to which command? |
| 07:55.39 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:55.45 | ChannelZ | you must wait for a response from a command |
| 07:56.47 | ChannelZ | and also * can tell you things at any time, it's not a 1-to-1 relationship - IE send a command, get a response. For instance if you use the Originate command, * might send you 3 or 4 responses showing the call progression |
| 07:56.55 | MAbbas | so essentially .. I can send only one command at a time .. |
| 07:57.07 | *** join/#asterisk lenne_dk (n=Leif@cpe.atm2-0-74391.hknxx4.customer.tele.dk) |
| 07:57.33 | ChannelZ | if you care about the responses yes |
| 08:00.45 | ChannelZ | You should probably read the responses anyway, if only to throw them away, as I'm not sure if the stdout would eventually clog up |
| 08:01.45 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 08:02.52 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 08:03.28 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 08:11.51 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 08:13.13 | MAbbas | ChannelZ: One problem, if I send a command and wait for response, meanwhile e.g. some events are fired by AM .. how do I know that the response does not belong to my command's response? |
| 08:14.37 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 08:17.34 | ChannelZ | I think an actual response will contain a Response: line whereas other random things * might say to you do not |
| 08:17.53 | ChannelZ | (the 'random things' would be Event: lines) |
| 08:18.32 | ChannelZ | http://www.voip-info.org/wiki/view/Asterisk+manager+API documents how most commands will respond |
| 08:20.06 | kaldemar | MAbbas: you can also put Events: off to login, and you don't have to worry about them. |
| 08:20.33 | MAbbas | But, on TCP level, if command's response is in mutiple packets, not every packet will contain the "Response" string .. |
| 08:20.40 | ChannelZ | that too |
| 08:22.00 | ChannelZ | You must read from the socket in whatever buffer sizes you're reading and assemble a 'complete message' yourself based on the CRLFCRLF.. |
| 08:22.21 | MAbbas | thats good workaround .. But in my case I need events too .. |
| 08:22.56 | ChannelZ | it's fairly common in socket programming, having to read chunks of data and parse/assemble it as determined by the protocol you're using, etc. |
| 08:23.43 | MAbbas | yes, thats the way to go .. only if turn off * events .. |
| 08:33.25 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 08:34.07 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:34.46 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 08:35.05 | *** join/#asterisk TimToady_ (n=moi@77.49.49.66.dsl.dyn.forthnet.gr) |
| 08:42.17 | *** join/#asterisk ticoit (n=ticoit@201.191.190.238) |
| 08:51.58 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:53.27 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 08:54.13 | ChannelZ | ok, so in > 1.4.24 it seems like app_voicemail.so requires res_smdi.so - fine.. but if I put preload => res_smdi.so as the first thing in modules.conf, or try to load => res_smdi.so and then load => app_voicemail.so, it all still fails |
| 08:55.09 | *** join/#asterisk tzafrir__laptop (n=tzafrir@local.xorcom.com) |
| 08:56.26 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 09:00.23 | *** join/#asterisk TommyBotten (n=noosjent@services.nbi.freecode.no) |
| 09:02.26 | ChannelZ | argh nevermind |
| 09:04.49 | *** join/#asterisk chendy (n=chatzill@116.24.38.61) |
| 09:24.16 | *** join/#asterisk ickmund (n=magnus@ada-bcn-fw01.adamoeurope.com) |
| 09:25.45 | TSM2 | is there anyway to have a beep to indicate that a atten transfer has been completed, we have an issue where users cant tell once the other party is now connected |
| 09:32.24 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:34.00 | *** join/#asterisk smooth_penguin (n=smoove@59.96.229.169) |
| 09:38.51 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 09:52.56 | *** join/#asterisk ticoit (n=ticoit@201.191.190.238) |
| 09:57.45 | *** join/#asterisk ruied (n=ruied@89.214.176.221) |
| 09:58.51 | *** join/#asterisk friehmaen (i=freeman@xers.de) |
| 10:00.49 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk StevenR (n=foo@wan1.sghs.org.uk) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk yoruk_ (n=yoruk@host164-182-dynamic.52-82-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 10:00.49 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM] |
| 10:04.34 | Tech_Travis | I'm running 1.6.0 at the office and wanted to test 1.6.1.1.11 in a virtual machine so I just installed it. I found it's missing SIP, IAX2, etc. and was wondering if that is normal or if I screwed something up in the install. |
| 10:05.43 | Tech_Travis | s/1.6.1.1.11/1.6.1.11 |
| 10:07.06 | *** join/#asterisk friehmaen (i=freeman@xers.de) |
| 10:16.03 | TimToady_ | by 'missing' you mean the modules, the configs, somehting else? |
| 10:17.05 | Tech_Travis | TimToaddy: inside the CLI running help doesn't show any of the SIP commands like sip show peers or sip show channels. |
| 10:17.40 | TimToady_ | make sure u have a sip.conf and then run 'module load chan_sip.so' |
| 10:18.21 | Tech_Travis | TimToady: okay, I'll go try now. |
| 10:18.26 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk StevenR (n=foo@wan1.sghs.org.uk) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk yoruk_ (n=yoruk@host164-182-dynamic.52-82-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 10:18.27 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM] |
| 10:20.37 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:20.41 | Tech_Travis | TimToady: There is no sip.conf in /etc/asterisk |
| 10:21.13 | TimToady_ | create one with ur settings |
| 10:21.31 | TimToady_ | or if you installed asterisk from source eun make samples for the sample conf files to get installed |
| 10:21.43 | TimToady_ | s/eun/run |
| 10:24.47 | *** join/#asterisk shinao1 (n=shinao1@41.219.214.230) |
| 10:28.24 | Tech_Travis | TimToady: Thanks for your help. I installed the sample files, rebooted, and now the SIP and IAX options are listed. I'm guessing this means that since there were no conf files when asterisk first installed it didn't load any of the modules, but now that conf files are there it's loading them? |
| 10:28.39 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 10:29.17 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 10:29.44 | TimToady_ | yes |
| 10:30.39 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 10:35.02 | Tech_Travis | TimToady: So by reverse logic on my old install I can remove conf files for things I'm not using like skinny and on the next restart * won't bother loading the associated module(s)? |
| 10:35.49 | kaldemar | that's bad practice. asterisk will try to load the module and you'll get errors and warning for missing configuration files. |
| 10:35.55 | TimToady_ | that does not aply for all modules, some can work without conf files. Best way is to check modules.conf and load only the modules you need |
| 10:36.22 | kaldemar | it you don't need a module, either delete the module itself or add a noload => <module_name>.so in /etc/asterisk/modules.conf |
| 10:36.23 | TimToady_ | by default asterisk tries to load every module (autoload=yes) |
| 10:38.33 | *** join/#asterisk ticoit (n=ticoit@201.191.190.238) |
| 10:43.52 | *** join/#asterisk mr_ian_ (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 10:44.46 | Tech_Travis | I would like to do the best practice approach, so along those lines is it suggested to not load modules that aren't used? Or should things be left stock unless there is a reason to deviate? |
| 10:47.18 | *** join/#asterisk yoshx (n=yoshx@78.114.253.27) |
| 10:48.17 | *** join/#asterisk nfi|ermes (n=ErMeS@host54-26-static.93-94-b.business.telecomitalia.it) |
| 10:49.13 | TimToady_ | unload everyhting you are not going to use, its the best and safest approach |
| 10:49.46 | TimToady_ | its just a bit hard to get it right the first time because of the number of the modules and that dependencies between some of them |
| 10:50.24 | TimToady_ | at least unload the channels that you are not going to use. |
| 10:51.34 | TommyBotten | I like the opposite, though similar approach. Autoload = no, and then load those thoungs you need. |
| 10:51.48 | TommyBotten | It is, as TimToady_ says - a bit hard the first time, choosing the modules you need. |
| 10:54.37 | Tech_Travis | This is a bit daunting the first time, which is why I'm testing this stuff in a virtual machine rather than the company's actual phone server. |
| 10:55.45 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk StevenR (n=foo@wan1.sghs.org.uk) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk yoruk_ (n=yoruk@host164-182-dynamic.52-82-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 10:55.45 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM] |
| 10:56.29 | Tech_Travis | Is there a command I can run or a directory I can look in to see what is available to turn off? |
| 10:57.25 | TimToady_ | 'modules show' in asterisk cli and ls -l /usr/lib/asterisk/modules/ |
| 10:57.34 | TSM2 | is there anyway to have a beep to indicate that a atten transfer has been completed, we have an issue where users cant tell once the other party is now connected |
| 10:58.33 | TommyBotten | Just do "for i in $(ls /usr/lib/asterisk/modules/); do echo "load =>" $i; done" |
| 10:58.46 | TommyBotten | And then pipe it to the modules.conf file |
| 11:01.42 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 11:02.18 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 11:09.32 | *** join/#asterisk viraptor (n=viraptor@212.11.80.34) |
| 11:10.10 | viraptor | him is there any asterisk-addons download page on the new * website? |
| 11:10.16 | viraptor | s/him/hi |
| 11:10.33 | Tech_Travis | TimToady & TommyBotten & kaldemar: Thanks for your time and insight I appreciate your help. Have a good evening (or morning depending on your timezone). |
| 11:10.47 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 11:12.49 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 11:12.58 | _cgc | hi everyone |
| 11:13.48 | _cgc | i have a couple of asterisk 1.6.1.8 server and keep getting the following warnings: chan_dahdi.c:10642 pri_fixup_principle: Call specified, but not found? chan_dahdi.c:11780 pri_dchannel: Hangup on bad channel 0/1 on span 1, is this anything to worry about? |
| 11:14.32 | TimToady_ | viraptor http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 11:15.38 | viraptor | TimToady_: yeah - I know that from google, but it's not linked from the website in any way, is it? |
| 11:16.31 | TimToady_ | you want to download the addons or to point that new page page sucks? :P |
| 11:17.30 | *** join/#asterisk ticoit (n=ticoit@201.191.190.238) |
| 11:19.13 | viraptor | TimToady_: I prefer: politely inform people who can change the page about a missing element, without using words - WTF, failure, suck and similar - political correctness, you see ;) |
| 11:19.52 | _cgc | any help on these warnings? chan_dahdi.c:10642 pri_fixup_principle: Call specified, but not found? chan_dahdi.c:11780 pri_dchannel: Hangup on bad channel 0/1 on span 1, i did a search on google but could only find a patch for version 1.6.0.5 |
| 11:21.13 | TimToady_ | viraptor you spoil the fun :P |
| 11:22.08 | *** join/#asterisk spiegel (n=spike@ip-166-187.interbild.net) |
| 11:26.26 | viraptor | TimToady_: you learn that kind of expressions when your company redesigns the main site and using "new website" and "sucks" in the same sentence becomes a sackable offence :D |
| 11:26.46 | lost_soul | lol |
| 11:27.00 | TimToady_ | ah so you are a web designer :P |
| 11:27.09 | lost_soul | I'll have to remember that, thanks viraptor |
| 11:27.36 | viraptor | :) |
| 11:28.04 | viraptor | anyways, just wanted to report that missing, c ya all |
| 11:28.06 | *** part/#asterisk viraptor (n=viraptor@212.11.80.34) |
| 11:30.57 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:42.18 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:42.18 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 11:42.18 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) |
| 11:42.18 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 11:42.18 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 11:42.18 | *** join/#asterisk StevenR (n=foo@wan1.sghs.org.uk) |
| 11:42.18 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 11:42.18 | *** join/#asterisk yoruk_ (n=yoruk@host164-182-dynamic.52-82-r.retail.telecomitalia.it) |
| 11:42.19 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 11:52.43 | *** join/#asterisk Kchehab (n=kchehab@212.98.141.202) |
| 11:53.38 | Kchehab | i am running asterisk 1.6.0.13 and patch from asterisk is available |
| 11:53.45 | Kchehab | to to apply the patch |
| 11:53.55 | Kchehab | http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ |
| 11:57.33 | *** part/#asterisk bbt (n=sam@2406:a000:f000:1400:0:0:0:1) |
| 12:03.49 | *** join/#asterisk [01]DND (n=arabia@80.227.221.34) |
| 12:04.11 | [01]DND | hi guys, is my plantronics h141N headset compatible with snom 360? |
| 12:04.31 | *** join/#asterisk qdk (n=qdk@0x573c2220.bynqu1.dynamic.dsl.tele.dk) |
| 12:07.11 | kaldemar | [01]DND: you didn't google much did you? |
| 12:12.07 | *** part/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 12:13.13 | [01]DND | actually i did. |
| 12:13.52 | [01]DND | but in the snom website it says its not compatible. but i also saw in the plantronicas website that there's an interchangeable cords. |
| 12:14.01 | [01]DND | *plantronics |
| 12:15.53 | *** join/#asterisk Assuero (n=Assuero@187.59.195.146) |
| 12:17.14 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 12:17.48 | ghenry | hi, how can you get the callerid via the Manager API of a SIP extension. I'm using SIPShowPeer |
| 12:18.04 | ghenry | Sipshowpeer rather on Asterisk 1.4 |
| 12:18.07 | ghenry | and get Callerid: "device" <502> |
| 12:18.12 | ghenry | as it's a FreePBX box |
| 12:18.27 | ghenry | I guess I have to map this via MySQL to a extension name |
| 12:18.35 | ghenry | join #freepbx |
| 12:21.07 | *** join/#asterisk zorp75ck (n=zorp75ck@pool-72-72-193-90.altnpa.east.verizon.net) |
| 12:28.26 | *** join/#asterisk _bugz_ (n=bugz@adsl-76-194-244-221.dsl.lsan03.sbcglobal.net) |
| 12:30.21 | angryuser | have someone successfully configured 1.6.0.10 and WANPIPE Release: 3.5.8.5 ? daemons are running, calls are initiated but somehow sangomas gateway does not care at all, incoming call don't even change state of bri lines, i ma sure that my T0 are working |
| 12:31.01 | angryuser | tested with another working system, and advice ? |
| 12:31.05 | angryuser | any* |
| 12:31.22 | angryuser | tzafrir, have you had any exp with sangomas ? |
| 12:31.53 | tzafrir | angryuser, not much |
| 12:32.35 | angryuser | i am sure about my bri setup, but something deep inside going wrong.. |
| 12:38.06 | tzafrir | do you use dahdi or their own stack? |
| 12:42.11 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 12:47.13 | *** join/#asterisk ruied (n=ruied@89.214.38.95) |
| 12:53.13 | *** join/#asterisk Madnashua (n=madnashu@78.147.124.54) |
| 12:58.02 | *** join/#asterisk Davedan (n=Administ@kaplun01.tau.ac.il) |
| 12:58.09 | Davedan | does asterisk support SIMPLE? |
| 12:58.46 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 12:59.37 | mbrevda | can I see the amount of registered licences from the cli? |
| 12:59.45 | kaldemar | Davedan: not all of it. be more precise. |
| 13:00.32 | Davedan | kaldemar: I need presence, contacts and group management |
| 13:00.47 | Davedan | kaldemar: add contact, remove contact, create a group, invite users to a group |
| 13:04.16 | kaldemar | asterisk is lacking those. |
| 13:06.05 | tzafrir | (it does support a different and much simpler presense mechanism in SIP: publish/subscribe) |
| 13:06.05 | Davedan | kaldemar: isn't there a way to define a roster like in IM? |
| 13:06.16 | tzafrir | (but I suspect this is not what you're after) |
| 13:07.01 | Davedan | tzafrir: I'm looking for an experience similar to XMPP or skype |
| 13:07.10 | kaldemar | Davedan: no. |
| 13:10.57 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 13:11.03 | killfill | hi |
| 13:11.21 | killfill | Queue(90|||4) <-- that should timeout after 4 seconds right? and should continue the dialplan? |
| 13:12.02 | kaldemar | in newer versions, you must replace | with a comma. |
| 13:12.10 | killfill | ah no its 1.4 |
| 13:12.16 | kaldemar | and timeout is the fourth parameter |
| 13:12.23 | kaldemar | core show application Queue |
| 13:14.01 | killfill | ah.. :) missed one |
| 13:15.52 | mpe | where is it posible to download the old asteriskNOW that is using asterisk gui and not freePBX |
| 13:16.54 | Pan3D | heh |
| 13:16.58 | Pan3D | morning |
| 13:20.29 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:22.41 | TommyBotten | I've set a few variables using SETVAR in sip.conf. Is it possible accessing these when calling *to* the user/peer instead of from? |
| 13:23.09 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
| 13:23.48 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 13:30.15 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:35.28 | dlynes | TommyBotten: You shouldn't be using SetVar() anymore...that's been deprecated as of 1.4, and probably obsoleted as of 1.6 |
| 13:35.43 | dlynes | TommyBotten: Use Set(...) instead |
| 13:36.30 | dlynes | TommyBotten: but, how do you propose to get the variables for the to, unless you're using a parameter to dial that dumps you into another context after the dial() has been completed? |
| 13:36.49 | dlynes | mpe: why would you want to? |
| 13:38.03 | dlynes | kaldemar: fwiw, I've found out there's an option you can specify in the general section to override the '|' / ',' behaviour....the '|' can still be used...just by default it's not allowed now |
| 13:38.09 | mpe | need to test a feature a customer of mine have, and the customer CD is 800miles away |
| 13:38.21 | dlynes | mpe: ah |
| 13:38.40 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:40.19 | ManxPower-work | TommyBotten didn't read UPGRADE*.txt |
| 13:40.30 | TommyBotten | Yes I did |
| 13:40.48 | TommyBotten | The setvar / set is fine. |
| 13:40.59 | TommyBotten | But that is not where the real issue is |
| 13:41.03 | ManxPower-work | TommyBotten: ALL of them? The deprecation of SetVar should have been documented |
| 13:41.27 | dlynes | mpe: what version number? |
| 13:41.48 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 13:42.03 | mpe | 2.0 |
| 13:42.21 | mpe | no i must be version 1 |
| 13:42.21 | TommyBotten | dlynes: Not really sure. In essence, what I'm trying to do is tie a secretary extension and a cellphone number to a SIP user. Any ideas? |
| 13:42.36 | tzafrir | mpe, where did you see this version number? |
| 13:42.55 | mpe | the one that supportet cisco phones |
| 13:42.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:44.13 | mpe | well the costomer is running on a closed system, so it is phone suport and he is a realy n00b, but I install the about a 1 / 1.5 year agow |
| 13:44.50 | mpe | and he only need to add a extra phone |
| 13:45.06 | ManxPower-work | TommyBotten: Ah, yes setvar= in sip.conf. No you cannot set those vars for calls TO the device. Only calls FROM the device. |
| 13:45.32 | mpe | but that is realy dificult to guide him without access to the gui |
| 13:45.55 | [TK]D-Fender | mpe: So go get access to the GUI... and this is not a GUI support channel |
| 13:46.31 | mpe | at the moment i'm trying to make a clean asterisk install and manualt instal asteriskgui 2.0 |
| 13:46.34 | TommyBotten | ManxPower-work: Hmm..Ok. But as I mentioned to dlynes, my task is really to tie a secretary and a cellphone to a SIP user. Any ideas on that? I have internal BDB or an external mysql ... but it doesn't "feel right" ;) |
| 13:47.00 | ManxPower-work | TommyBotten: I have no idea what yo mean "tie a secretary and cell phone to a sip user" |
| 13:47.09 | [TK]D-Fender | TommyBotten: Ame here.... explain... |
| 13:47.11 | [TK]D-Fender | Same* |
| 13:47.17 | mpe | on a spare server, just wanto to save time by reusing the same asteriskNOW image |
| 13:48.40 | [TK]D-Fender | mpe: AsteriskNOW doesn't come with AsteriskGUI any more. The old one was based on rPath+AsteriskGUI and loses you a lot of support options |
| 13:48.57 | ManxPower-work | filthy GUIs |
| 13:50.07 | mpe | yes, but that is the one I need, as that is what the customer have |
| 13:50.47 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 13:51.42 | ManxPower-work | mpe: and yet you are asking on this non-GUI channel. |
| 13:52.12 | ManxPower-work | makes about as much sense as asking Windows95 questions on a DOS channel. |
| 13:53.58 | TommyBotten | ManxPower-work / [TK]D-Fender : The users have a set of secretaries - also defined in the same config. I would like to have a mapping that connects or ties a SIP user (end user) to another SIP user (secretary) |
| 13:54.21 | ManxPower-work | TommyBotten: Exactly what do you want to accomplish with that "mapping" |
| 13:54.22 | [TK]D-Fender | TommyBotten: tie how? With rope? |
| 13:54.59 | TommyBotten | ManxPower-work / [TK]D-Fender : From a functional perspective: When DND is enabled, only the secretary may call the user. And when external calls are not answered, it should be forwarded to the users secretary. |
| 13:55.01 | tzafrir | mpe, if you have a question about asterisk, ask here. If you have a question about the asterisk-gui, ask on #asterisk-gui |
| 13:55.02 | ManxPower-work | TommyBotten: It's starting to sound like you are looking for a Key System, not a PBX. |
| 13:55.08 | *** join/#asterisk hatpanda (n=pete-joh@triton.dsv.su.se) |
| 13:55.22 | hatpanda | Anyone using ser together with asterisk? |
| 13:55.27 | mpe | sorry but was just looking for the iso , and the gui channel is practaly empty |
| 13:55.29 | ManxPower-work | TommyBotten: should not be hard to do at all |
| 13:55.43 | ManxPower-work | mpe: that empty channel should tell you something. |
| 13:55.44 | tzafrir | mpe, what ISO? |
| 13:55.48 | [TK]D-Fender | mpe: The ISO will probably be hard to find now. |
| 13:55.53 | ManxPower-work | tzafrir: the asterisk iso! |
| 13:56.02 | tzafrir | *.iso |
| 13:56.03 | [TK]D-Fender | tzafrir: *NOw (with AsteriskGUI+rPath) |
| 13:56.17 | [TK]D-Fender | TommyBotten: Basic dialplan |
| 13:56.32 | tzafrir | AsteriskNoW is no longer based on rPath, actually |
| 13:56.48 | mpe | yes but I need the oldverion :--( |
| 13:56.49 | [TK]D-Fender | tzafrir: Which we also told him |
| 13:57.01 | TommyBotten | [TK]D-Fender / ManxPower-work: Shouldn't be hard no. But I think I'm about to create a corner case. Any tips would be very much appreciated |
| 13:57.14 | tzafrir | mpe, the old version will have asterisk-gui 1.x |
| 13:57.23 | [TK]D-Fender | TommyBotten: Corner case? How many books does that hold? |
| 13:57.28 | tzafrir | Is this what you need? |
| 13:57.32 | mpe | yes |
| 13:57.34 | ManxPower-work | TommyBotten: call comes into "boss" extension, check DND and the source |
| 13:57.45 | dlynes | mpe: I've tried a number of combinations for the version number on the download server, but nothing's coming up except the current version |
| 13:57.51 | mpe | need for a support case of mine |
| 13:58.00 | dlynes | mpe: why didn't you save a copy of your original download? |
| 13:58.19 | ManxPower-work | dlynes: people do that? 8-| |
| 13:58.26 | TommyBotten | ManxPower-work: That is what I'm doing. But where do I define which number is allowed to call or not? Doing this for one or two is easy. I have 300 mappings to do |
| 13:58.32 | dlynes | ManxPower-work: I always do |
| 13:58.40 | TommyBotten | ManxPower-work: Doing this in dialplan manually will be really ugly |
| 13:58.41 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:58.41 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 13:58.41 | dlynes | ManxPower-work: I've got downloads up to about 2 or 3 years ago |
| 13:58.43 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:58.45 | mpe | yes i also wonder way I dident save the file |
| 13:58.49 | ManxPower-work | TommyBotten: you did not say "300" mappings. You said a secretary and a boss. |
| 13:59.02 | dlynes | ManxPower-work: especially for important stuff like asterisk and mozilla |
| 13:59.06 | ManxPower-work | TommyBotten: you would want to use some form of database. |
| 13:59.07 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 13:59.14 | ariel_ | Morning folks |
| 13:59.29 | ManxPower-work | dlynes: but every version of asterisk ever released is still available. |
| 13:59.32 | mpe | just got used to redowloading as i have 100/100 mbit |
| 13:59.43 | dlynes | ManxPower-work: who's to say they will be in the future? |
| 13:59.47 | [TK]D-Fender | TommyBotten: So 1 minor DB. Big deal. |
| 14:00.01 | TommyBotten | ManxPower-work: Makes sense. Is this a typical internal BDB or external stuff? |
| 14:00.02 | dlynes | ManxPower-work: asterisknow is case in point |
| 14:00.17 | [TK]D-Fender | TommyBotten: there is no such thing as "typical" |
| 14:00.26 | ManxPower-work | dlynes: I never considered AsteriskNOW to be Asterisk. I also never considered it a core Digium product. |
| 14:01.00 | mpe | yes asteriskNow is a strange beast |
| 14:01.04 | TommyBotten | [TK]D-Fender: You know what I mean. |
| 14:01.12 | ManxPower-work | dlynes: I see your point, however. I do wonder why people keep trying to use AsteriskNOW. Nobody seems to support it on IRC, it keeps going thru major changes, and apparently you can't even get old versions anymore. |
| 14:01.23 | [TK]D-Fender | TommyBotten: With your wording I'm not going to start guessing... |
| 14:01.39 | [TK]D-Fender | TommyBotten: I could picture you using either |
| 14:01.39 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 14:02.05 | dlynes | ManxPower-work: who knows? you're better off to just install centos, and then install asterisk and freepbx from source |
| 14:02.28 | dlynes | ManxPower-work: or maybe even install pbx in a flash |
| 14:02.47 | TommyBotten | [TK]D-Fender: English is not my mother tongue, hence the confusion. But thanks for your insight. |
| 14:02.50 | ManxPower-work | dlynes: or just install Asterisk and stop trying to make a Fisher Price PBX |
| 14:02.56 | TommyBotten | ManxPower-work: Thanks to you aswell |
| 14:03.02 | mpe | yes but I only need it so I can make support for a customer that is suing the version |
| 14:03.17 | DocAwesome | get new customers :) |
| 14:03.20 | dlynes | ManxPower-work: heh...first time i've heard it called that, but yeah...good description :) |
| 14:03.34 | dlynes | mpe: i'd sue that version, too |
| 14:03.55 | mpe | all my curent customers now have asterisk from svn 1.4 |
| 14:03.55 | [TK]D-Fender | litigates |
| 14:03.56 | ManxPower-work | Most of our customers have a "GUI"fied version of Asterisk. As far as I can tell they are just as stupid when using the GUI as they are when trying to use a CLI. |
| 14:04.18 | tzafrir | mpe, I figure you can still download it from rPath: http://www.rpath.org/project/asterisk/release?id=5501 |
| 14:04.24 | tzafrir | Up-to-date it isn't |
| 14:04.29 | dlynes | ManxPower-work: Yeah..I've noticed the same thing...even with their so-called administrators |
| 14:04.31 | mpe | but som old have asterisk@home or asteriskNOW |
| 14:04.57 | ManxPower-work | dlynes: Yup. So really the GUI solves almost no problems and causes plenty of other problems. |
| 14:04.58 | mpe | thanks just the link i needed |
| 14:05.33 | Kchehab | i am running asterisk 1.6.0.13 and patch from asterisk is available to v 1.6.0.13 |
| 14:05.43 | Kchehab | how to apply the patch |
| 14:05.56 | Kchehab | patch found here http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ |
| 14:05.59 | dlynes | ManxPower-work: yeah...the problem we had after my boss mandated freepbx, was that we couldn't use alphanumeric SIP usernames...they had to be numeric |
| 14:06.22 | ManxPower-work | dlynes: same applies to extensions in FreePBX |
| 14:06.25 | dlynes | ManxPower-work: which sucked big time when the facility decided to start calling all the rooms Waterfront Shores South, ... |
| 14:06.36 | dlynes | ManxPower-work: instead of B wing N |
| 14:06.39 | dlynes | or whatever |
| 14:06.53 | dlynes | erm B Wing, 3rd Floor |
| 14:06.57 | Katty | :< |
| 14:07.05 | DocAwesome | Kchehab: cd /usr/src/asterisk-1.6.0.13 ; patch -p0 < the_patch_you_downloaded |
| 14:07.06 | Katty | i must face reality and accept it is now winter. /tear |
| 14:07.16 | ManxPower-work | I normally use the MAC of the device as it's SIP username. |
| 14:07.20 | Kchehab | @DocAwesome thanks |
| 14:07.24 | dlynes | because for taht, I could just use a simple extension 2304 for B Wing 3rd Floor, room 4....but i'd be lost when they gave me their new stupid floor names |
| 14:07.37 | Kchehab | @DocAwesome i shlould recompile asterisk |
| 14:07.51 | dlynes | ManxPower-work: yes, but I get told to hook up a certain room number |
| 14:07.56 | DocAwesome | Kchehab: beyond that, the exercise is for you to figure it out as this is not a #linux support channel |
| 14:08.23 | dlynes | ManxPower-work: It's much easier if everything's already self-documented, so I don't need to cross reference with a spreadsheet, or something similar |
| 14:08.26 | [TK]D-Fender | [09:04]<mpe>but som old have asterisk@home or asteriskNOW <_ A@H is ANCIENT |
| 14:08.56 | DocAwesome | dlynes: tying devices to a particular location or extension number is not the best idea |
| 14:09.07 | ManxPower-work | [TK]D-Fender: once someone exploits a security issue with that old Asteirsk I'm sure someone will start thinking maybe they are screwed. |
| 14:09.11 | dlynes | DocAwesome: It's all internal (not external access) |
| 14:09.13 | DocAwesome | those items should be abstracted away from the device |
| 14:09.29 | DocAwesome | dlynes: what if someone needs to move a phone to a new location because some other device broke? |
| 14:09.37 | DocAwesome | then your system in asterisk is immediately wrong |
| 14:09.49 | dlynes | DocAwesome: they won't...they're all hard-wired into the building infrastructure |
| 14:10.00 | DocAwesome | do whatever you want |
| 14:10.06 | dlynes | DocAwesome: it's all 24 pair fxs mediatrix gateways |
| 14:10.42 | dlynes | DocAwesome: so if something fails, I swap out one 4124, and swap in a new one |
| 14:10.49 | DocAwesome | ok |
| 14:10.58 | *** part/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.56 | dlynes | This policy of not showing when people log in or log out kinda sucks |
| 14:12.30 | dlynes | ManxPower-work: isn't that what I just said? about the numeric extensions in freepbx? |
| 14:12.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:12.58 | ManxPower-work | dlynes: no you said something about locations. |
| 14:13.23 | ManxPower-work | I'm using the one thing that specifies the device in a globally unique way. 8-) |
| 14:13.30 | dlynes | [09:05]<dlynes>ManxPower-work: yeah...the problem we had after my boss mandated freepbx, was that we couldn't use alphanumeric SIP usernames...they had to be numeric |
| 14:13.51 | ManxPower-work | (9:06:25 AM) dlynes: ManxPower-work: which sucked big time when the facility decided to start calling all the rooms Waterfront Shores South, ... |
| 14:13.59 | dlynes | Yes, but then you need a spreadsheet to cross reference everything |
| 14:14.11 | ManxPower-work | why? |
| 14:14.19 | dlynes | Besides, when I've got 24 sip extensions all with the same mac address, how does that help? |
| 14:14.44 | *** join/#asterisk zorp75ck (n=zorp75ck@pool-72-72-193-90.altnpa.east.verizon.net) |
| 14:14.48 | ManxPower-work | dlynes: I use -a -b -c postfixes to indicate the specific line. |
| 14:15.15 | [TK]D-Fender | DNYmAYBE YOU COULD USE YOUR BRAIN AND COME UP WITH SOME THING SLIGHTLY better |
| 14:15.16 | dlynes | ManxPower-work: ah...so like macaddr-waterfrontshoressouth304 |
| 14:15.22 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) |
| 14:15.23 | [TK]D-Fender | Gah caps.... |
| 14:15.24 | ManxPower-work | but I've never used a "sip fxs gateway". |
| 14:15.35 | ManxPower-work | dlynes: no like macaddr-a |
| 14:15.49 | ManxPower-work | If I want analog I use a channel bank. |
| 14:16.11 | dlynes | ManxPower-work: yes, because you're using pris everywhere, if i remember correctly |
| 14:16.22 | dlynes | ManxPower-work: we're not using pris at all...pure sip trunks |
| 14:16.28 | ManxPower-work | dlynes: no. |
| 14:17.08 | dlynes | ManxPower-work: oh...you're using analog lines, too? |
| 14:18.03 | ManxPower-work | dlynes: at my new job we mostly use SIP for everything -- except for analog ports. FAX analog goes on POTS and is not connected to the PBX, if we other analog we use an analog card or a channel bank |
| 14:18.23 | dlynes | ManxPower-work: ah |
| 14:21.11 | dlynes | [TK]D-Fender: yeah...I suppose I could use astdb to do a mapping between the ports, or something, and just create some kind of gui for administering the relationship |
| 14:21.18 | dlynes | [TK]D-Fender: that talks to ami |
| 14:21.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:21.55 | [TK]D-Fender | dlynes: Umm... why? |
| 14:22.19 | [TK]D-Fender | dlynes: Mapping between what and what? |
| 14:24.46 | dlynes | [TK]D-Fender: these assinine floor names, and room numbers and mac addresses/port numbers |
| 14:25.18 | ManxPower-work | dlynes: what would you use the info for? |
| 14:25.24 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 14:25.53 | ManxPower-work | if all else fails you can use setvar= in sip.conf |
| 14:25.58 | dlynes | [TK]D-Fender: so that when they email saying they need Waterfront Shores South, Room 4 hooked up, I could just program that into the gui, and then have it make the association that that's Mac address B800009C2A, port 23 |
| 14:26.09 | ManxPower-work | setvar=LOCATION=3rd floor mens room |
| 14:26.15 | [TK]D-Fender | dlynes: just make the user an arbitrary common prefix + floor + room |
| 14:26.46 | ManxPower-work | dlynes: ah, you work for idiots. I get e-mails saying "provision MAC as extension 25" |
| 14:26.56 | ManxPower-work | I don't CARE where it is. 8-) |
| 14:27.19 | dlynes | ManxPower-work: They have enough issues telling me Building B, Room 304, and they can't handle that anymore |
| 14:27.55 | dlynes | ManxPower-work: I can't see them possibly asking me Mac address xxxx port 23 |
| 14:28.06 | _abc_ | hello, any problems with running 1.2.13 instead of latest when expecting to use skinny or sccp ? |
| 14:28.16 | _abc_ | i need to test something without upgrading my machine |
| 14:28.25 | dlynes | ManxPower-work: these aren't who I work for...they're customers....nurses |
| 14:28.29 | ManxPower-work | dlynes: port? I can see that in analog, but not for SIP |
| 14:28.36 | [TK]D-Fender | dlynes: So make a speadsheet and stop making it sound like sucha big deal |
| 14:28.51 | dlynes | ManxPower-work: SIP trunk -> mediatrix 4124 24 port fxs gateway -> analog phones |
| 14:28.52 | [TK]D-Fender | ManxPower-work: .... 24-port SIP GATEWAYS <- |
| 14:29.34 | dlynes | [TK]D-Fender: i like whining, and I don't want to make a full time job just taking care of one facility |
| 14:29.43 | *** join/#asterisk jmworx__ (n=jeval@mail.octasic.com) |
| 14:29.44 | [TK]D-Fender | _abc_: That version is no longer supported and hos more security holes than a #9 sponge, but by all means... |
| 14:29.52 | ManxPower-work | [TK]D-Fender: yeah, but I don't consider people that use SIP gateways to be "real people" |
| 14:30.05 | _abc_ | [TK]D-Fender: ok, i just need it to test configs not on the net |
| 14:30.15 | dlynes | [TK]D-Fender: the amount of work I'm having to do just for one care facility is more than ten times what I need to do for the other facilities |
| 14:30.18 | _abc_ | ManxPower-work: so what are they ... sippers ? |
| 14:30.19 | [TK]D-Fender | dlynes: You should pick your clients better then... and/or find another therapist. You aren't paying us enough for all of this ;) |
| 14:30.43 | [TK]D-Fender | ManxPower-work: Sure they are, and I recommend them over channel banks myself |
| 14:30.51 | dlynes | [TK]D-Fender: I have no otion in the customer picking process |
| 14:30.51 | ManxPower-work | _abc_: "people that like to make stuff more complicated than they have to be" |
| 14:31.19 | [TK]D-Fender | _abc_: It'll probably work the same now as it did then. |
| 14:31.23 | _abc_ | so any problems with 1.2.13 i should be aware of? remember this is for sip/skinny/sccp configuration testing, offline. there are no security concerns |
| 14:31.54 | russellb | um, yes? |
| 14:31.54 | [TK]D-Fender | dlynes: Then go get another therapist... |
| 14:31.54 | _abc_ | mkay, i asked |
| 14:31.54 | russellb | _abc_: that release is insanely old |
| 14:31.54 | dlynes | [TK]D-Fender: yeah...the mediatrix 4124's I find are considerably cheaper and more flexible than the channel banks |
| 14:31.58 | _abc_ | well its the stock on debian etch which is what i have available to test with |
| 14:32.01 | beek | infobot tell me about nat |
| 14:32.19 | dlynes | [TK]D-Fender: besides...they're a hell of a lot more useful than those damned audiocodes boxes (and way better support) |
| 14:32.40 | [TK]D-Fender | dlynes: I'm a short drive away from their head-office |
| 14:32.48 | _abc_ | russellb: so is there anything i should watch out for? as in, setup incompatible and such? |
| 14:32.51 | beek | morning [TK]D-Fender |
| 14:32.52 | dlynes | _abc_: why on earth would you use a binary distribution of asterisk, if you don't have to? |
| 14:33.03 | russellb | _abc_: well, there have been a bunch of security fixes since then. |
| 14:33.03 | ManxPower-work | I bet your "sip channel bank" lets you label ports -- which is where they should be labeled |
| 14:33.08 | [TK]D-Fender | _abc_: If its the same version, why are you expecting differences? |
| 14:33.09 | dlynes | _abc_: debian's known for not keeping releases within the last decade |
| 14:33.11 | russellb | They may have been backported ... |
| 14:33.14 | _abc_ | dlynes: because my primary goal is to debug setups before i put them on another machine which runs a more modern version |
| 14:33.24 | russellb | I should stop trying to jump in to the middle of conversations. |
| 14:33.28 | russellb | wanders off |
| 14:33.34 | dlynes | russellb: heh |
| 14:33.46 | ManxPower-work | _abc_: very few people use prepackaged Asterisk. |
| 14:34.04 | _abc_ | well its convenient, i just get the debs and it works |
| 14:34.18 | _abc_ | the only thing i care about now is whether the setups are portable |
| 14:34.23 | dlynes | _abc_: debian's release policy works well for most stuff, but it's not terribly practical for asterisk |
| 14:34.25 | _abc_ | skinny sip and sccp confs |
| 14:34.30 | ManxPower-work | _abc_: until you get hacked or crash, or find that nobody will help you with such an old version |
| 14:34.51 | dlynes | _abc_: the setups are portable (if both setups are the same version) |
| 14:34.58 | _abc_ | well if need be i'd upgrade. my system is so old i need to upgrade serious libs to compile modern stuff on it |
| 14:35.05 | tzafrir | ManxPower-work, or you rebuild a deb version |
| 14:35.16 | _abc_ | dlynes: you mean from 1.2.13 to 1.6.x ? |
| 14:35.21 | [TK]D-Fender | ~abcif the versions are going to be different you are asking for trouble and show no concept of understanding the scientific process |
| 14:35.22 | infobot | [TK]D-Fender: okay |
| 14:35.32 | dlynes | _abc_: yes....or even 1.4.24 to 1.4.26 you might break something |
| 14:35.38 | [TK]D-Fender | _abc_: if the versions are going to be different you are asking for trouble and show no concept of understanding the scientific process |
| 14:35.43 | tzafrir | _abc_, 1.2.13? please upgrade from oldstable :-) |
| 14:35.56 | dlynes | _abc_: but guaranteed, 1.2.13 to 1.6.x you're going to break something for sure |
| 14:35.59 | _abc_ | funny i started with linux kernel 1.2.13 now it's back to 1.2.13 with asterisk |
| 14:36.04 | dlynes | _abc_: they're completely incompatible |
| 14:36.12 | _abc_ | ok, how do you know this? |
| 14:36.21 | dlynes | _abc_: kernel 1.2.13 was on debian 0.1 beta |
| 14:36.34 | _abc_ | dlynes: i had it on slackware 3.3 i think |
| 14:36.37 | tzafrir | _abc_, do you plan on upgrading to Lenny any time soon? |
| 14:36.50 | _abc_ | tzafrir: nah, why break it if it works ;) |
| 14:37.04 | dlynes | tzafrir: he's not even on etch, I don't think, if he's still running Linux 1.2.13 kernel |
| 14:37.08 | tzafrir | because pretty soon there won't be any security fixes for it |
| 14:37.29 | _abc_ | i'm on etch people, ASTERISK is 1.2.13, ok? my kernel is 2.6.30 something |
| 14:37.47 | _abc_ | please read what i write not what you think i wrote ;) |
| 14:37.55 | dlynes | _abc_: so stop mixing up your terms...you said you were on Linux kernel 1.2.13 |
| 14:38.41 | _abc_ | no i did not, read the backlog |
| 14:39.16 | _abc_ | enough circus, time for bread. has anyone had to edit configs in sip skinny or sccp when upgrading asterisk? recently? at any time? |
| 14:39.21 | [TK]D-Fender | _abc_: What is your deployment version going to be? |
| 14:39.36 | _abc_ | [TK]D-Fender: probably the latest stable asterisk |
| 14:40.14 | ManxPower-work | [TK]D-Fender: too bad he's setting himself to fail. |
| 14:40.41 | _abc_ | yes, baby, YES, give it to me. now the clue-stick: WHY would i fail? |
| 14:40.52 | kaldemar | _abc_: yes. read UPGRADE*.txt in the source package to know what to edit. |
| 14:41.06 | ManxPower-work | _abc_: because you will have to rewrite your configs when moving from 1.2.x to 1.6.x.x |
| 14:41.46 | _abc_ | aha, so i would need to read all the UPGRADE*.txt files in all the asterisk packages since 1.2.13 to now to catch problems? is there a place where these are available in bulk so i can grab them and grep them? |
| 14:42.02 | ManxPower-work | _abc_: all the upgrade files are in all the versions of asterisk |
| 14:42.24 | Katty | brr. |
| 14:42.26 | _abc_ | ManxPower-work: ok, so what kind of rewrite are we talking about, small details or total rewrites from scratch |
| 14:42.49 | ManxPower-work | _abc_: I am not going to spend a couple hours tutoring you on something you can go read yourself. |
| 14:43.03 | _abc_ | of course not, thanks for the hints |
| 14:43.05 | *** join/#asterisk skirmisha (n=ast@87-126-34-92.btc-net.bg) |
| 14:43.12 | skirmisha | hi guys |
| 14:43.15 | Katty | hi |
| 14:43.20 | *** part/#asterisk Madnashua (n=madnashu@78.147.124.54) |
| 14:43.27 | skirmisha | any ideas how to solve problem with ast multiple ips |
| 14:43.34 | [TK]D-Fender | _abc_: Ok. we're going to stop now. These versions are too differrent. Massive changes between these 2. |
| 14:43.56 | ppc | yo |
| 14:43.57 | skirmisha | i send to second ip and i got reply from first |
| 14:44.07 | skirmisha | is there solution for that |
| 14:44.10 | _abc_ | ok, thanks for the tips. am going to get and compile the latest stable one now. wish me luck |
| 14:44.15 | skirmisha | both ips are on same card |
| 14:44.23 | ManxPower-work | skirmisha: the source is determined by your OS routing table. |
| 14:44.47 | skirmisha | yes i saw that written somewhere |
| 14:44.52 | skirmisha | but it does not make sense to me |
| 14:45.06 | skirmisha | because ips are on same card |
| 14:45.13 | skirmisha | eth0 and eth0:0 |
| 14:45.59 | skirmisha | so how can i set ast to answer with proper ip |
| 14:46.15 | _abc_ | would you people recommend an asterisknow install instead of compilation? |
| 14:46.18 | skirmisha | and another thing is it possible asterisk to listen on multiple ports? |
| 14:46.26 | [TK]D-Fender | _abc_: No |
| 14:46.33 | ManxPower-work | _abc_: we wouldn't. people on #AsteriskNOW might. |
| 14:46.34 | [TK]D-Fender | skirmisha: Yes |
| 14:46.37 | _abc_ | why? it looks simpler/faster |
| 14:46.43 | skirmisha | how? |
| 14:47.01 | ManxPower-work | skirmisha: see sip.conf.sample and rtp.conf.sample for bindip/bindaddr |
| 14:47.03 | [TK]D-Fender | _abc_: Because it DOESN"T install the "latest" version, and includes a GUI that gets in the way |
| 14:47.12 | _abc_ | i see |
| 14:47.31 | skirmisha | bindip get first variable defined in config and ingnore second one |
| 14:47.36 | skirmisha | i tested it already |
| 14:47.47 | [TK]D-Fender | skirmisha: you made the WRONG CHOICE |
| 14:47.47 | *** join/#asterisk MarkJenks (i=ce2871c7@gateway/web/freenode/x-rxpvwdktzplzblph) |
| 14:47.52 | skirmisha | how about multiple ports |
| 14:47.52 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 14:47.56 | *** join/#asterisk RypPn2 (n=Gloop@rossgroup2.demon.co.uk) |
| 14:48.02 | [TK]D-Fender | skirmisha: Binds to 1 or ALL. |
| 14:48.05 | ManxPower-work | skirmisha: first you talk about IPs now you talk about ports. |
| 14:48.08 | [TK]D-Fender | skirmisha: So leave it alone. |
| 14:48.17 | _abc_ | is there a dependency list for 1.6 ? |
| 14:48.27 | skirmisha | i want either ips or ports |
| 14:48.28 | [TK]D-Fender | _abc_: go read the docs. |
| 14:48.38 | _abc_ | meh. thanks anyway |
| 14:48.39 | skirmisha | i tried both but it does not work with the config |
| 14:48.43 | ManxPower-work | [TK]D-Fender: this is looking like a Monday |
| 14:48.51 | skirmisha | curretly i have bindaddr 0.0.0.0 |
| 14:49.12 | skirmisha | and set 2 ips on the eth0 |
| 14:49.25 | skirmisha | but second ip packet's answer with first ip |
| 14:49.48 | skirmisha | give me some solution that has been tested |
| 14:49.53 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 14:50.06 | skirmisha | otherwise it is just talking |
| 14:50.28 | skirmisha | when i add to bindaddr fields it reads first met |
| 14:50.34 | skirmisha | second is ignored |
| 14:50.41 | skirmisha | with ports i think is same |
| 14:50.44 | [TK]D-Fender | skirmisha: So it IS listening on that port and yous till don't understand how your OS's routing table picks the interface * is going to REspOND with |
| 14:51.16 | ManxPower-work | UGh. Verizon is bouncing lines all over the place. |
| 14:51.16 | [TK]D-Fender | skirmisha: ManxPower-work mentioned this to you already but you don't seem open to learning your networking implications. Good luck with that. |
| 14:51.30 | [TK]D-Fender | skirmisha: And dual(+)-homes * = PITA |
| 14:51.42 | ManxPower-work | [TK]D-Fender: It's a PBX, you don't need to know networking! |
| 14:51.49 | ppc | haha |
| 14:52.00 | skirmisha | let me read over |
| 14:52.01 | [TK]D-Fender | [09:50]<skirmisha>when i add to bindaddr fields it reads first met <- ONE or ALL. What part did you miss? |
| 14:53.33 | tzafrir | [TK]D-Fender, off-topic: just that you know what 'PITA' reminds me: http://duckduckgo.com/?q=PITA |
| 14:54.26 | skirmisha | [TK]D-Fender don't get you |
| 14:54.55 | [TK]D-Fender | tzafrir: I'm well aware of its myriad meanings, and that site sucks :) |
| 14:55.26 | tzafrir | Actually, I'm using it now as my main search engine |
| 14:57.32 | [TK]D-Fender | tzafrir: Go right ahead.... |
| 14:58.11 | skirmisha | ok here is the situation i have now. * binds to all and i have 2 ips. I understand routing table of ast and that's why it picks first ip. How can i tell * to pick proper ip then? |
| 14:58.30 | skirmisha | routing table of OS i meant |
| 14:58.38 | ManxPower-work | skirmisha: you basically cannot tell Asteirsk to pick the "proper ip" |
| 14:59.04 | tzafrir | skirmisha, you want to use one IP for SIP phones in the LAN and a different one for SIP phones outside it? |
| 14:59.04 | skirmisha | ok that part is clear |
| 14:59.27 | skirmisha | no i want to have multiple ports on sip |
| 14:59.34 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 14:59.37 | skirmisha | i use 2 ips but sending on diff port |
| 14:59.41 | skirmisha | and to port forward |
| 14:59.41 | fcois93 | hello all |
| 14:59.45 | skirmisha | do i meant |
| 14:59.49 | Faustov | sip bindaddr option lets pick the proper ip |
| 15:00.02 | Faustov | if you put that in [general] |
| 15:00.07 | Faustov | under sip.conf |
| 15:00.19 | skirmisha | yes currently is bind to all |
| 15:00.31 | skirmisha | this is not the issue |
| 15:00.36 | Faustov | the default might be 0.0.0.0 = bind all |
| 15:00.40 | ManxPower-work | skirmisha: what you are trying to do is an advanced networking issue. |
| 15:00.47 | skirmisha | is it possible to set more than 1 list port? |
| 15:01.06 | skirmisha | ManxPower-work yes i know |
| 15:01.13 | skirmisha | i have load balance infront of it |
| 15:01.14 | Faustov | skirmisha: tried 5060,5061 <--- something like this? |
| 15:01.21 | fcois93 | I cant read a sip header which was insert by another asterisk! I have the problem when the asterisk A insert an header. if asterisk B insert an header the next asterisk can read it! why can't I read the header? |
| 15:01.25 | ManxPower-work | skirmisha: why are you not asking on a networking channel? |
| 15:01.41 | ManxPower-work | fcois93: does the header show up in sip debug? |
| 15:01.50 | skirmisha | because it is ast related issue |
| 15:01.59 | ManxPower-work | skirmisha: no. it. is. not. |
| 15:02.01 | voipmonk | skirmisha, you choose the ip you want by setting the bindaddr in sip.conf - if you want more than one ip you will need to look in virtualization or kamailio |
| 15:02.17 | *** join/#asterisk wierdo (n=jimmy@89.252.206.114) |
| 15:02.24 | voipmonk | if I understand your questions and your goal... |
| 15:02.30 | [TK]D-Fender | voipmonk: Sure, why don't you jsut HAND him the answer ! :p |
| 15:02.32 | fcois93 | ManxPower-work: yes it is in the sipdebug |
| 15:02.42 | Faustov | voipmonk: I've just suggested that to him :D |
| 15:02.46 | [TK]D-Fender | swats voipmonk |
| 15:02.49 | voipmonk | takes the battery out of the SpoonFeed 2.0 |
| 15:02.53 | voipmonk | sorry |
| 15:02.54 | voipmonk | :) |
| 15:02.59 | *** join/#asterisk MAbbas (i=Jinbaba@115.186.15.133) |
| 15:03.01 | [TK]D-Fender | ups the amperage on SpoonFeed 2.0 |
| 15:03.06 | voipmonk | ahhh |
| 15:03.10 | [TK]D-Fender | FRY BABY FRY! |
| 15:03.15 | voipmonk | u let the magic smoke out |
| 15:03.17 | voipmonk | :( |
| 15:03.24 | voipmonk | smells burnt silicon |
| 15:03.34 | [TK]D-Fender | voipmonk: Yup.. computers runn on smoke... when they release theirs... they stop working! |
| 15:03.45 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:04.24 | kaldemar | fcois93: by now you should know to pastebin configs and a CLI output for a call and include a link in the question. |
| 15:04.33 | skirmisha | will this work with double ports separate by coma |
| 15:04.40 | skirmisha | 5060,5061 |
| 15:04.49 | skirmisha | have not tested that yet |
| 15:04.49 | ManxPower-work | skirmisha: does it say it will in sip.conf.sample? |
| 15:04.58 | Faustov | skirmisha: inmany examples it does |
| 15:05.01 | Faustov | try. |
| 15:05.36 | skirmisha | it is not in example config |
| 15:05.38 | Faustov | also, you could redirect the traffic from one port to another with iptables if you were really desperate |
| 15:06.14 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 15:06.22 | ManxPower-work | skirmisha: then I doubt you can do that |
| 15:06.26 | skirmisha | yes i am doing port forward, but only in one way |
| 15:06.48 | Faustov | you can do it both ways with dnat and snat |
| 15:06.52 | Faustov | pretend there are 2 ports |
| 15:07.27 | voipmonk | good lord |
| 15:07.40 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:07.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:07.48 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:08.47 | eppigy | NEIN |
| 15:09.03 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 15:09.15 | fcois93 | kaldemar, ManxPower-work : http://pastebin.com/d395e9c96 |
| 15:09.39 | ManxPower-work | I'm starting to suspect that every dyslexic in the world work for telecos. |
| 15:10.37 | _abc_ | ManxPower-work: why yuo men? |
| 15:10.55 | fcois93 | kaldemar ManxPower-work: I try to read 'x-conf_room-room' for example. it works with others servers. but when the server which send that invite insert headers, I can't read it! it is an asterisk |
| 15:10.56 | dlynes | tzafrir: so, is it possible to bind to one ip address for one sip trunk, and a different ip address for another sip trunk like skirmisha was asking, then? I've been curious about that lately, myself |
| 15:11.16 | *** join/#asterisk Akiraa (n=Akira@79.112.30.9) |
| 15:11.17 | _abc_ | you have to be dislexic to stay in telecom with the crisis ... no-one else would take you eh ? |
| 15:11.24 | _abc_ | ducks and runs away |
| 15:11.58 | MAbbas | In my dialplan I use AGI script to route incoming call to a particular agent(using "Dial"), If agi is failed to route I add call to a queue for default processing. |
| 15:12.06 | ManxPower-work | dlynes: no such thing as SIP Trunk. You should be able to have different user/friend listen on a specific port, but that's almost always totally useless. |
| 15:12.38 | dlynes | ManxPower-work: i was thinking more like peer |
| 15:13.05 | tzafrir | dlynes, bind to all, but use localnet and externip |
| 15:13.16 | kaldemar | fcois93: the X headers should start with a capital X. |
| 15:13.26 | ManxPower-work | dlynes: peers make OUTGOING calls has nothing to do with the incoming listen port. |
| 15:13.28 | kaldemar | fcois93: by all means, also show an attempt to read them. |
| 15:13.39 | dlynes | ManxPower-work: ok, so i got the terms backwards |
| 15:13.48 | MAbbas | Is there any way, using AMI I get controll of the call and route it. Instead of adding it to Queue for default processing? |
| 15:14.19 | fcois93 | I test... |
| 15:14.19 | dlynes | ManxPower-work: What I want is two have two virtual ips...one will take ulaw calls from one external ip, and the other will take g729 calls from a different external ip |
| 15:14.38 | ManxPower-work | dlynes: why do it that way? |
| 15:14.51 | [TK]D-Fender | MAbbas: MAbbas Get control of what? |
| 15:14.51 | fcois93 | but if it is a SER which insert a small x-... no problems... |
| 15:14.58 | dlynes | ManxPower-work: is there another way, without having to run two separate boxes, or two separate asterisk installs? |
| 15:15.18 | voipmonk | MAbbas, yes |
| 15:15.41 | MAbbas | [TK]D-Fender: controll of call routing .. as AGI script get control of dialplan execution .. |
| 15:15.52 | ManxPower-work | dlynes: I used to do that all the time. two peer/friend/users each with different codecs allowed. |
| 15:15.54 | [TK]D-Fender | MAbbas: You aren't making any sens |
| 15:15.56 | [TK]D-Fender | e |
| 15:16.19 | dlynes | ManxPower-work: ok...let me go one step further...without usernames/passwords? |
| 15:16.28 | kaldemar | fcois93: SER does what it is told to. |
| 15:16.55 | fcois93 | kaldemar: the 'X-...' don't work |
| 15:16.58 | ManxPower-work | dlynes: if you are not using usernames and passwords then you are beyond even my help (and I suspect [TK]D-Fender's too) |
| 15:17.19 | MAbbas | [TK]D-Fender: dialplan waits for AGI script to complete and then proceeds further .. AMI being event based... Can I pause dailplan execution while AMI is making a decision regarding call routing |
| 15:17.27 | fcois93 | kaldemar: if I insert the header with the SER it don't work! I have some problem with the asterisk sender I think |
| 15:17.30 | dlynes | ManxPower-work: ok...so separate box, or separate instance of asterisk then bound to a different static address |
| 15:17.33 | kaldemar | fcois93: you're not showing what the thing is that should work in the first place. |
| 15:20.05 | [TK]D-Fender | MAbbas: AMI can redirect a call, but I don't know how that will work if the call is in an AGI |
| 15:20.15 | *** join/#asterisk codefreeze-lap (n=murf@12.130.118.76) |
| 15:20.28 | [TK]D-Fender | MAbbas: Your goal is unclear as are the terms under which this hijacking is to occur |
| 15:21.03 | [TK]D-Fender | fcois93: Do you see * adding the header? |
| 15:22.29 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 15:22.46 | fcois93 | [TK]D-Fender: yes |
| 15:23.05 | fcois93 | [TK]D-Fender: I see the headers in the sip debug in the asterisk destination |
| 15:23.10 | kaldemar | fcois93: why are we not seeing it? |
| 15:23.25 | MAbbas | let try to be more clear .. A calls comes in, using AMI is it possible to route call to an agent? if the route not successful, handle the call in its dialplan context. |
| 15:23.33 | fcois93 | kaldemar: I see the first asterisk insert the headers |
| 15:23.43 | [TK]D-Fender | MAbbas: Wht is AMI involved at all? What the hell is your DIALPLAN doing? |
| 15:23.48 | fcois93 | kaldemar: the second can't read headers wich are here! |
| 15:23.52 | [TK]D-Fender | MaWhat are these decisions based on? |
| 15:24.00 | [TK]D-Fender | MAbbas: Please find a clue, and find it fast... |
| 15:24.04 | kaldemar | fcois93: again, why are we not seeing the whole picture? why is it not in a pastebin? |
| 15:24.08 | ManxPower-work | fcois93: are you using different versions of Asterisk? |
| 15:24.22 | fcois93 | ManxPower-work: same evrsion |
| 15:24.34 | [TK]D-Fender | fcois93: ........... |
| 15:24.43 | [TK]D-Fender | fcois93: you don't seem to be listening... |
| 15:24.44 | ManxPower-work | fcois93: then I guess the output of sip debug and the output of a failed call is what is needed. put it on pastebin |
| 15:25.08 | fcois93 | my problem is that I have more than 4 asterisk. I have that problem between 2 asterisk |
| 15:25.18 | fcois93 | ok |
| 15:25.24 | ManxPower-work | fcois93: no. Your problem is lack of pastebin |
| 15:25.52 | ManxPower-work | It's a good thing we can wait around all day and keep asking for the info. |
| 15:25.56 | fcois93 | ManxPower-work: in the first asterisk, I have -- Executing [1234567@testeur:10] SIPAddHeader("SIP/ser_conf-b755b170", "X-conf_room-room: 1234567") in new stack |
| 15:26.31 | ManxPower-work | fcois93: you are ignoring what I am saying. last chance for a pastebin before I write you off as an idiot. |
| 15:26.36 | [TK]D-Fender | fcois93: Stop with the worthless sotry and pastebin the CODE, and the CALL |
| 15:27.14 | fcois93 | ManxPower-work: the second can't read if I do exten => _X.,n,NoOp(room:${SIP_HEADER(X-conf_room-room)}) |
| 15:27.23 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:27.30 | MAbbas | let say, a calls comes in dialplan context and I invoke an AGI script which tries to route call to particular agent. If routing met an error, I add this call to queue. Can same scenario be achieved with AMI instead of using AGI? |
| 15:27.32 | ManxPower-work | Ah well. I wish you the BEST of luck, fcois93. |
| 15:28.26 | [TK]D-Fender | MAbbas: What error? Why the hell is AMI involved? Why isn't your AGI making the decisions? |
| 15:28.44 | [TK]D-Fender | MAbbas: And stop saying "route". |
| 15:29.18 | [TK]D-Fender | MAbbas: AMI does not process calls. DIALPLAN does. AGI is a way to run dialplan apps from withing a script coded in a language of yuor choosing |
| 15:29.34 | *** join/#asterisk jonavogt (n=jonavogt@u51-229.dsl.vianetworks.de) |
| 15:30.01 | MAbbas | lets say in AGI I try to transfer call to AGent/2001 which is busy hence the error .. |
| 15:30.27 | ManxPower-work | MAbbas: exactly what do you do in your AGI to "transfer" a call? |
| 15:30.40 | Kobaz | MAbbas: so then you check DIALSTATUS and process accordingly |
| 15:30.41 | [TK]D-Fender | MAbbas: IF THE DIAL FALLS THROUGH THEM agi CAN CONTINUE DOING WHATEVER YOU WANT. YOU DON'T NEED ami FOR ANYTHING |
| 15:30.46 | MAbbas | I call "Dial" application .. |
| 15:30.56 | [TK]D-Fender | MAbbas: Dial returns a result code. |
| 15:31.01 | ManxPower-work | MAbbas: then say "dial". "transfer" is totally different. |
| 15:31.02 | Kobaz | if ${DIALSTATUS} == "BUSY" |
| 15:31.06 | [TK]D-Fender | MAbbas: go LOOK AT IT |
| 15:31.21 | Kobaz | MAbbas: core show application dial |
| 15:31.23 | jonavogt | Hi, I got a problem using mISDN. Calling from external sources it symply won't generate any ringing tone unless I specify 'r' with dial. Any hints what is missing? |
| 15:31.30 | [TK]D-Fender | MAbbas: it is not a "route" or a "transfer". never use these terms for this again. |
| 15:31.38 | MAbbas | What if I dont want to use AGI, instead acheive samething using AMI.. Is it possible? |
| 15:32.25 | MAbbas | [TK]D-Fender: whats the appropriate term? |
| 15:33.09 | dlynes | MAbbas: you could try using a combination of AMI and call files...that could be a possibility |
| 15:33.25 | [TK]D-Fender | MAbbas: Its a friggen DIAL command. |
| 15:33.36 | [TK]D-Fender | dlynes: Please don't confuse him further |
| 15:33.53 | dlynes | [TK]D-Fender: well, call files can do the dial comamnd |
| 15:33.58 | dlynes | s/comamnd/command/ |
| 15:34.00 | [TK]D-Fender | MAbbas: AMI does not fucking process calls. Are we clear? DIALPLAN processes calls. |
| 15:34.26 | [TK]D-Fender | dlynes: He's jsut processing a call. |
| 15:34.52 | dlynes | [TK]D-Fender: ah...thought he was trying to do something complicated....guess he's trying to do something simple, but in a complicated way |
| 15:35.01 | MAbbas | What does "http://www.voip-info.org/wiki/view/Asterisk+manager+experience" means when the guys says .. "Redirect ANY live call to ANY destination, some examples" |
| 15:35.10 | [TK]D-Fender | dlynes: No.. he has no clue about Asterisk, and less communications skills |
| 15:35.31 | [TK]D-Fender | MAbbas: that is not step by step processing, it is a 1-shot REDIRECT |
| 15:35.35 | *** join/#asterisk ticoit (n=ticoit@190.241.180.89) |
| 15:35.47 | [TK]D-Fender | MAbbas: AMI is not for processing call-flow. NO STEP BY STEP |
| 15:35.54 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 15:36.24 | bcrisp | ::ping:: |
| 15:36.31 | dlynes | [TK]D-Fender: ah...i see what he's looking for now....yes....terribly simple....Answer()...Dial(...) |
| 15:36.44 | bcrisp | morning guys |
| 15:37.03 | dlynes | bcrisp: no, you can just download it from the same place you downloaded the asterisk source...no need for svn |
| 15:37.21 | dlynes | bcrisp: http://downloads.digium.com/pub/telephony/asterisk/ |
| 15:37.32 | bcrisp | yeah i ended up getting it.. just wanted to make sure i had the proper version |
| 15:37.49 | bcrisp | everything installed and now i got dahdi installed/meetme working which is great |
| 15:37.57 | fcois93 | [TK]D-Fender: ManxPower-work: kaldemar: http://pastebin.com/d3df35045 there are destination and from |
| 15:38.03 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:38.03 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:38.07 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:38.07 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 15:38.14 | *** join/#asterisk heliosj (n=jeff@207.140.8.67.cfl.res.rr.com) |
| 15:38.39 | [TK]D-Fender | fcois93: Where's the code? |
| 15:39.03 | fcois93 | [TK]D-Fender: you mean the dialplan? |
| 15:39.21 | [TK]D-Fender | fcois93: YES. We've asked for all of this a dozen times. |
| 15:40.13 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
| 15:40.30 | bcrisp | in order to change the "comedian mail" i just overwrite the default vm-login sound correct? |
| 15:40.45 | [TK]D-Fender | fcois93: And show us the ENTire incoming call |
| 15:40.50 | [TK]D-Fender | bcrisp: Yes |
| 15:40.51 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 15:42.28 | bcrisp | downloads audacity |
| 15:42.51 | fcois93 | [TK]D-Fender: here the part of dialplan |
| 15:43.19 | fcois93 | [TK]D-Fender: up there is another pastbin with sipdebug |
| 15:45.02 | *** join/#asterisk codefreeze-lap (n=murf@12.130.118.76) |
| 15:45.15 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
| 15:46.31 | *** join/#asterisk codefreeze-lap (n=murf@12.130.118.76) |
| 15:47.00 | *** join/#asterisk wopsy (n=MEGAWOPS@cap31-3-82-227-199-59.fbx.proxad.net) |
| 15:50.17 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 15:50.56 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:51.37 | dlynes | bcrisp: congratulations |
| 15:51.55 | bcrisp | heh |
| 15:52.27 | dlynes | bcrisp: see? it wasn't as big of a deal as you made it out to be, was it? |
| 15:52.48 | bcrisp | nah it wasnt too bad |
| 15:52.59 | bcrisp | reading the linux book helped a bit |
| 15:53.04 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.53.211) |
| 15:53.28 | bcrisp | i think the issue before was the fact that the kernel was updated but the headers were old |
| 15:53.32 | bcrisp | this time i didnt update the kernel |
| 15:54.01 | bcrisp | but you're right, should be easier to troubleshoot without plesk madness installed |
| 15:55.23 | dlynes | bcrisp: it's just another layer of bullshit that's not needed...myself, I don't have a choice...I'm stuck with it because we're using 1 and 1, and we need a gui for domain management so that my boss can manage it, another office staffer can manage it, and all of our customers can do their own minor management |
| 15:55.25 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:55.43 | bcrisp | in that sense it seems ok |
| 15:56.07 | dlynes | bcrisp: that being said, cpanel and virtualmin/webmin/usermin pretty much do the same thing, but virtualmin's not as complete |
| 15:56.20 | dlynes | bcrisp: not sure about cpanel, as I've never used it |
| 15:56.41 | dlynes | bcrisp: and hsphere's another beast that's more or less the same thing, but way more complicated than plesk |
| 15:56.44 | *** join/#asterisk wopsynator (n=MEGAWOPS@cap31-3-82-227-199-59.fbx.proxad.net) |
| 15:57.01 | MAbbas | <dlynes> and [TK]D-Fender: http://www.voip-info.org/boards/index.php?t=18865 |
| 15:57.25 | bcrisp | dlynes do you know of any simple software for visualizing queue information - learning AMI at the moment but is taking some time |
| 15:57.58 | dlynes | MAbbas: what determines what the 'best agent' is? |
| 15:58.07 | voipmonk | MAbbas , you want a skill based routing call center system..... this doesnt require AGI |
| 15:58.42 | MAbbas | I am using an AI algorithm to decide that .. |
| 15:58.42 | [TK]D-Fender | and AGI is not a background process |
| 15:58.47 | dlynes | MAbbas: it would seem to me what you need is just a simple dialplan, some queues, and maybe some astdb foo |
| 15:59.01 | [TK]D-Fender | MAbbas: Let us know when you've got regular intelligence figured out... |
| 15:59.14 | Katty | how about i steam some squirrels. |
| 15:59.24 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:59.28 | MAbbas | voipmonk: its not skill based routing .. within a skill .. I further catogarize .. |
| 15:59.29 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:59.36 | dlynes | Katty: sounds good...can you have it ready in about an hour? I'd love some for lunch... |
| 15:59.40 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 15:59.46 | [TK]D-Fender | MAbbas: And you are not going to completely hijack the agent selection method for Queue externally like that, so forget it. |
| 15:59.57 | [TK]D-Fender | MAbbas: "vi app_queue.c" <---- |
| 16:00.10 | voipmonk | MAbbas: why didnt you add that to the requirements? |
| 16:00.31 | dlynes | voipmonk: because he wants us all to waste our time |
| 16:00.32 | [TK]D-Fender | voipmonk: So this story can get better EVERY TIME HE TELLS IT! |
| 16:00.57 | MAbbas | <[TK]D-Fender>: plz stop mocking .. I am new to asterisk .. so :) |
| 16:01.05 | dlynes | voipmonk: just like every freepbx user that comes in here, and doesn't bother telling us they're using freepbx until 1/2 hour later |
| 16:01.33 | ManxPower-work | dlynes: Those people should be smacked over the head with a big bat made of doam. |
| 16:01.39 | ManxPower-work | foam, even |
| 16:01.45 | [TK]D-Fender | MAbbas: "vi app_queue.c" <---- |
| 16:01.46 | dlynes | ManxPower-work: hehe...no kidding |
| 16:01.58 | voipmonk | I want a blue car..... ok here's a blue car..... no I want one with run flats.....ok here's one with runflats.... no I need one with a twin turbo..... ok here's one with twin turbo...... no I want one thats electric and the twin turbo is for a blow dryer for my wife for the back seat.... ok here's the price and the waiting list for that one...... I dont want a waiting list, I want it now.... and I want it free... ok give me $20 for a carton of bullets. |
| 16:01.58 | voipmonk | I dont want a ...... here.... I'll give you $20 ( pulls trigger ) |
| 16:02.10 | ManxPower-work | Thank *smack* you *smack* for *smack* wasting *smack* our *smack* time |
| 16:02.28 | voipmonk | LOL!!!!!!!!! |
| 16:03.05 | [TK]D-Fender | voipmonk: http://tinyurl.com/yow2q8 |
| 16:03.11 | ManxPower-work | What's really sad is when Asterisk newbies try to do advanced PBX design and call routing. |
| 16:03.27 | dlynes | MAbbas: instead of wasting everyone's time, do up a flowchart, so we have some kind of clue what you need (AND DON'T LEAVE ANYTHING OUT) |
| 16:03.36 | dlynes | ~pics |
| 16:03.37 | infobot | pics is probably http://pics.bzleague.com/ |
| 16:03.37 | DocAwesome | I don't want to know your name! I just want... |
| 16:03.41 | Katty | infobot: critter cam |
| 16:03.42 | infobot | rumour has it, critter cam is http://www.ustream.tv/channel/squirrel-critter-cam |
| 16:03.52 | [TK]D-Fender | dlynes: NO NEED |
| 16:03.52 | Katty | ^- now streaming. |
| 16:03.58 | [TK]D-Fender | dlynes: Again you are way behind the curve |
| 16:04.04 | DocAwesome | ~blc |
| 16:04.08 | [TK]D-Fender | doc! ! ! |
| 16:04.09 | dlynes | [TK]D-Fender: ? |
| 16:04.14 | DocAwesome | infobot: blc is Big League Chew |
| 16:04.15 | infobot | DocAwesome: okay |
| 16:04.23 | [TK]D-Fender | dlynes: [11:01]<[TK]D-Fender>MAbbas: "vi app_queue.c" <---- |
| 16:04.40 | ManxPower-work | ~answers |
| 16:04.41 | infobot | answers is, like, Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 16:04.41 | MAbbas | Guys .. sorry for wasting your time .. I will draw flow chart too .. |
| 16:04.42 | dlynes | [TK]D-Fender: if he's new to asterisk, he probably doesn't even know what a C compiler is, much less what vi is |
| 16:04.48 | [TK]D-Fender | MAbbas: NO NEED |
| 16:04.57 | [TK]D-Fender | MAbbas: pay attention |
| 16:05.00 | [TK]D-Fender | MAbbas: "vi app_queue.c" <---- |
| 16:05.01 | fcois93 | ManxPower-work: no answer for my problem ? |
| 16:05.16 | [TK]D-Fender | fcois93: of course not |
| 16:05.36 | dlynes | [TK]D-Fender: he could try writing some complicated dialplan, skip queues altogether, implementing his own queues in the dialplan, using Local channels, couldn't he? |
| 16:05.40 | ManxPower-work | fcois93: I told you that if you didn't provide the requested information I was going to write you off as an idiot. You did not provide the information before you asked several more questions. You are an idiot. |
| 16:05.40 | [TK]D-Fender | [11:04]<dlynes>[TK]D-Fender: if he's new to asterisk, he probably doesn't even know what a C compiler is, much less what vi is |
| 16:05.49 | [TK]D-Fender | dlynes: Dumb assumption |
| 16:05.59 | dlynes | [TK]D-Fender: why is it a dumb assumption? |
| 16:06.03 | MAbbas | <[TK]D-Fender>: "vi app_queue.c" will it work for different versions of asterisk? |
| 16:06.08 | [TK]D-Fender | dlynes: Being new to Asterisk does not mean new to programming or *NIX. |
| 16:06.20 | dlynes | [TK]D-Fender: see my point? He just proved it :) |
| 16:06.21 | [TK]D-Fender | dlynes: Look at these wild jumps you're making. |
| 16:06.47 | *** join/#asterisk Raszh (n=Spoon@12.185.1.72) |
| 16:06.49 | dlynes | [TK]D-Fender: he thinks 'vi' is an asterisk command |
| 16:06.49 | bcrisp | im a total linux newb.. found out about vi while reading the asterisk starter ebook |
| 16:07.15 | [TK]D-Fender | MAbbas: for a range of minor versions within a branch probably. Between majors... don't bet on it |
| 16:07.42 | MAbbas | no sir, I mean if I modify queue code .. will it work for all versions of * |
| 16:07.51 | dlynes | ah |
| 16:07.53 | [TK]D-Fender | MAbbas: I just answered you.... NO! |
| 16:08.14 | dlynes | MAbbas: it may not even work in the next minor revision of asterisk |
| 16:08.32 | dlynes | MAbbas: it may even break from one svn check in to the next |
| 16:08.33 | [TK]D-Fender | dlynes: probably should... |
| 16:08.45 | fcois93 | [TK]D-Fender: why an asterisk can't read headers? it can read from an asterisk but nott from another one! |
| 16:08.52 | [TK]D-Fender | fcois93: ... |
| 16:08.59 | [TK]D-Fender | ~wmmfpb ???? |
| 16:09.00 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 16:09.06 | dlynes | [TK]D-Fender: probably...but look at just the changes between 1.4.25 and 1.4.26 |
| 16:09.08 | bcrisp | lol |
| 16:09.20 | [TK]D-Fender | fdcoiWe asked you to show us the &*#ing code like 10 times. WAKE The HELL UP |
| 16:09.28 | bcrisp | is it just me, or does meetme rock? |
| 16:09.28 | fcois93 | [TK]D-Fender: I gave you all information look up |
| 16:09.41 | [TK]D-Fender | fcois93: No, you DIDN"T. you said loko up for it. IT WASN'T THERE |
| 16:10.02 | [TK]D-Fender | bcrisp: Its jsut you |
| 16:10.06 | [TK]D-Fender | bcrisp: sry |
| 16:10.21 | dlynes | [TK]D-Fender: he means 45 minutes ago, i think |
| 16:10.32 | dlynes | [TK]D-Fender: i.e. like about 6 screens back |
| 16:10.34 | [TK]D-Fender | dlynes: Not anywhere near the times we asked |
| 16:10.47 | bcrisp | [TK]D-Fender: you closet MeetMe lover... |
| 16:10.53 | MAbbas | So, whats the way to go then .. if it will not work with modifying .. queues code .. from one ver to another .. |
| 16:11.19 | dlynes | [10:37]<fcois93>[TK]D-Fender: ManxPower-work: kaldemar: http://pastebin.com/d3df35045 there are destination and from |
| 16:13.37 | ManxPower-work | MAbbas: the way to go is get good at using/managing/configuring Asterisk before you try rewriting it. |
| 16:13.37 | bcrisp | uh |
| 16:13.37 | [TK]D-Fender | MAbbas: for what you want to do you need to modify source. Get off your ass and deal with it |
| 16:13.37 | [TK]D-Fender | dlynes: ywheres the #&*$^ing DIALPLAN CODE in there? |
| 16:13.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:13.38 | dlynes | [TK]D-Fender: i have no idea...never clicked on the link |
| 16:13.38 | dlynes | [TK]D-Fender: i just figured that's what he was talking about, cause that's the only link he's ever posted |
| 16:13.38 | [TK]D-Fender | dlynes: You don't seem to be on the same planet as the rest of us... |
| 16:13.38 | kaldemar | dlynes: that's not what we asked him for ~10 times. |
| 16:13.38 | dlynes | kaldemar: yeah...so i've noticed...didn't look at the link before |
| 16:14.15 | dlynes | kaldemar: I just assumed he couldn't possibly be so daft |
| 16:15.07 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk keycruncher (n=dennis@pa-67-239-45-242.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 16:15.08 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk jblack (n=jblack@71.181.241.176) |
| 16:15.08 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 16:15.08 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 16:15.08 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk vlt (n=dm@suez.activ-job.com) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk pfn (n=pfnguyen@66.218.53.2) |
| 16:15.08 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk Chotaire (i=chotaire@to.to.to) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk shinao1 (n=shinao1@41.219.214.230) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk xrmx__ (n=rm@host143-252-dynamic.8-87-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk d00gster (n=doughant@94.98.134.132) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 16:15.08 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 16:15.08 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk rbd (n=rbd@rrcs-98-101-33-14.midsouth.biz.rr.com) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) [NETSPLIT VICTIM] |
| 16:15.08 | *** join/#asterisk RonDutt (n=aiontsu@69.7.43.20) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk jackal (n=jackal@pool-96-247-204-126.clppva.fios.verizon.net) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:15.09 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk ix33 (n=ix@52.48.4f.static.xlhost.com) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 16:15.09 | *** join/#asterisk n0cturnal (n=n0c@203.161.119.242.static.amnet.net.au) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) [NETSPLIT VICTIM] |
| 16:15.09 | *** join/#asterisk SuperBock (n=admin@mx-cln-1.netcanvas.com) [NETSPLIT VICTIM] |
| 16:15.10 | *** join/#asterisk Superbartt (n=bart@213.10.33.201) [NETSPLIT VICTIM] |
| 16:15.10 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 16:15.10 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
| 16:15.10 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) [NETSPLIT VICTIM] |
| 16:15.10 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 16:15.10 | *** join/#asterisk styelz (n=yoohoo@2001:5c0:1500:800:0:0:0:1) [NETSPLIT VICTIM] |
| 16:15.10 | bcrisp | whats up with the mass exodus |
| 16:15.11 | [TK]D-Fender | dlynes: You make a LOT of assumptions... the vast majority of them wrong. |
| 16:15.30 | *** join/#asterisk jtodd (i=csii1j32@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:15.30 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk heliosj (n=jeff@207.140.8.67.cfl.res.rr.com) |
| 16:15.30 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk spiegel (n=spike@ip-166-187.interbild.net) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk yoshx (n=yoshx@78.114.253.27) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 16:15.30 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk jmm3 (i=ident-us@gwyn.tux.org) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) |
| 16:15.30 | *** join/#asterisk cnu (i=cnu@113.80-203-44.nextgentel.com) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk nd- (n=nd-@xes-mad.com) |
| 16:15.30 | *** join/#asterisk mangala (i=mangala@antenora.aculei.net) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk zamba (i=marius@flage.org) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk p3nguin (i=BuhPjX1J@24-171-73-146.dhcp.mtvr.il.charter.com) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk McLazarus (n=McLazaru@dogpile.mcallister.ws) [NETSPLIT VICTIM] |
| 16:15.30 | *** join/#asterisk jksM (i=jks@193.189.93.254) [NETSPLIT VICTIM] |
| 16:15.31 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 16:15.31 | *** mode/#asterisk [+o jtodd] by irc.freenode.net |
| 16:15.32 | [TK]D-Fender | dlynes: *b00m* |
| 16:15.32 | voipmonk | i farted |
| 16:15.32 | bcrisp | lol |
| 16:15.47 | [TK]D-Fender | voipmonk: Geneva.... CONVENTION!!! *gasp* |
| 16:16.05 | Katty | oooh netsplitty |
| 16:16.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.27 | Deeewayne | watches Katty's squirrels |
| 16:16.41 | fcois93 | [TK]D-Fender: one more question: ${SIP_HEADER(x-conf_room-room)} read just the SDP ? because, I could see that asterisk don't insert in sdp but upper. SER insert in the SDP. when SER insert:no problem. if asterisk insert: problem |
| 16:16.53 | Katty | Deeewayne: not much happenin |
| 16:17.03 | [TK]D-Fender | fcois93: Where's the complete call? Where's the dialplan? |
| 16:17.05 | eppigy | negative squirell activity |
| 16:17.08 | Deeewayne | I know. He ran away |
| 16:17.10 | Deeewayne | or she |
| 16:17.20 | [TK]D-Fender | fcois93: STOP WASTING OUR TIME |
| 16:17.28 | Deeewayne | Katty, they are chasing each other around the tree! |
| 16:17.35 | Katty | they do that. often. |
| 16:17.36 | kaldemar | fcois93: asterisk handles SIP headers when it is told to, not SDP headers. |
| 16:17.41 | Katty | chickadee! |
| 16:17.44 | Katty | house wren! |
| 16:17.51 | Katty | chickadee! |
| 16:17.54 | eppigy | 8[] |
| 16:17.57 | Katty | cardinal! |
| 16:18.00 | Deeewayne | I wonder if I can encourage some to come into my neighborhood from the farm across the street |
| 16:18.10 | Katty | Deeewayne: yes. |
| 16:18.17 | fcois93 | [TK]D-Fender: http://pastebin.com/d3df35045 |
| 16:18.17 | fcois93 | [TK]D-Fender: http://pastebin.com/d43598b3f |
| 16:18.18 | Katty | Deeewayne: they travel great distances to get lunch. |
| 16:18.23 | dlynes | bcrisp: what mass exodus? |
| 16:18.50 | fcois93 | kaldemar: is it possible to insert in the SDP as SER can do ? |
| 16:18.59 | bcrisp | i saw ~ 50 users drop out of the channel |
| 16:19.02 | [TK]D-Fender | Ok, I'm done wasting time on this crap. |
| 16:19.09 | Katty | 4 house wrens (= |
| 16:19.10 | Katty | 5! |
| 16:19.10 | Deeewayne | are those croutons ? |
| 16:19.23 | Katty | croutons? ^_- |
| 16:19.24 | Deeewayne | gourmet squirrel food |
| 16:19.31 | Katty | those are banana slices |
| 16:19.35 | Deeewayne | oooh |
| 16:19.39 | Katty | and acorns...sunflower seeds, and corn |
| 16:19.57 | bcrisp | Katty: for future reference, i prefer apple slices |
| 16:20.05 | Katty | bcrisp: so do the squirrels. |
| 16:20.09 | Deeewayne | partay at Katty's lunch spot |
| 16:21.07 | eppigy | i liek dried apples |
| 16:21.24 | Katty | now i'mhungry |
| 16:21.43 | *** join/#asterisk Quasar-1922 (n=quasar@mail.kdj.nl) |
| 16:22.24 | Quasar-1922 | Hey everyone.. I'm having trouble with cmd Pickup. I'd like to pickup an incoming ZAP call that is in an extension listening to an IVR menu.. Is this possible?? |
| 16:22.43 | eppigy | me too |
| 16:23.12 | Katty | http://42ndrecipestreet.blogspot.com/2009/09/thai-peanut-chicken-and-noodles.html <- using turkey |
| 16:23.39 | ManxPower-work | Quasar-1922: you can only pick up RINGING calls. |
| 16:24.03 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 16:24.11 | Katty | hello mister pike |
| 16:26.22 | Quasar-1922 | ->Manxpower-work Ok, that's what I thought..is there a way to simulate a ringing call? |
| 16:26.38 | Katty | Poll Question: What's your favorite cold weather food? |
| 16:26.51 | Deeewayne | Pizza |
| 16:26.52 | DocAwesome | Katty: hot chocolate or lasagna |
| 16:27.04 | Deeewayne | Pizza is my favorite warm weather food too |
| 16:27.21 | coppice | casserolled lotus root |
| 16:27.22 | Deeewayne | specifically, NJ Pizza |
| 16:27.50 | ManxPower-work | w/ pepperoni and GREEN olives |
| 16:28.03 | Katty | you know...no one ever says soup |
| 16:28.09 | DocAwesome | I don't like olives on my pizza |
| 16:28.15 | DocAwesome | or tomatoes |
| 16:28.20 | DocAwesome | but I like them on salads and such |
| 16:28.24 | Katty | not a fan of olives myself, actually |
| 16:28.41 | eppigy | man i love olives |
| 16:28.47 | eppigy | mostly the green ones |
| 16:28.51 | Katty | eppigy: you're just weird. |
| 16:28.55 | eppigy | yesh |
| 16:28.57 | eppigy | it is true |
| 16:29.01 | Katty | we still love you. |
| 16:29.05 | Katty | even if you do like olives ^_- |
| 16:29.07 | eppigy | oh good :> |
| 16:29.33 | Katty | DocAwesome: do you have a lasagna recipe? |
| 16:29.35 | DocAwesome | I didn't used to like olives, but I've grown to enjoy them lately |
| 16:29.48 | DocAwesome | Katty: not really... but I like both veggie and meat based lasagnas |
| 16:30.03 | Katty | DocAwesome: where do you get lasagna then if you don't make it? |
| 16:30.05 | Deeewayne | mmmm.... lasagna .... |
| 16:30.07 | DocAwesome | as long as there is some rigotta (sp?) in it |
| 16:30.12 | DocAwesome | Katty: sometimes the store :) |
| 16:30.15 | DocAwesome | or the g/f makes it :) |
| 16:30.16 | DocAwesome | or my mom |
| 16:30.21 | Katty | oh i see. |
| 16:30.22 | Deeewayne | DocAwesome, a lasagna store ? |
| 16:30.32 | DocAwesome | Deeewayne: down in the lasagna district |
| 16:30.37 | Deeewayne | lol |
| 16:30.44 | Katty | i don't know of any places around here that have lasagna on their menu, actually |
| 16:30.51 | eppigy | :< |
| 16:30.59 | DocAwesome | Katty: where do you live, so I know not to move there |
| 16:31.08 | Katty | cape girardeau |
| 16:31.17 | Katty | bout 4 hours north west of huntsville |
| 16:31.17 | jaytee | coincidentally I'm making lasagne for a lunch tomorrow with coworkers. do you know how hard it is to find whole milk ricotta in this hick town? |
| 16:31.22 | eppigy | makes notes for stalking |
| 16:31.40 | DocAwesome | ah... ricotta, not rigotta |
| 16:31.41 | Katty | eppigy: you already have my phone number |
| 16:31.45 | eppigy | yesh |
| 16:31.47 | eppigy | this is true |
| 16:31.54 | Katty | how much more stalkery can there really be |
| 16:31.55 | eppigy | we should talk some time |
| 16:32.10 | Katty | i don't like talking |
| 16:32.21 | eppigy | haha yeah im not much of a phone talker either |
| 16:32.32 | [TK]D-Fender | Quasar-1922: Do an AMI Redirect on the call. |
| 16:32.32 | jaytee | plus I was shocked that the price of a 28oz can of San Marzano tomatoes costs $5.29 that's outrageous even if they are imported |
| 16:32.51 | Katty | jaytee: so you've noticed this trend of low fat and fat free cheese products too eh? |
| 16:33.01 | jaytee | yep |
| 16:33.05 | Katty | jaytee: it's a conspiracy. |
| 16:33.11 | Katty | jaytee: they're doing away with real food. |
| 16:33.18 | jaytee | and I never use part-skim mozzarella on lasagne or pizza either |
| 16:33.24 | eppigy | ugh |
| 16:33.27 | *** join/#asterisk chai_sangeen (n=chai_san@84.255.164.36) |
| 16:33.29 | Katty | i never use part-skim mozarella, period. |
| 16:33.38 | eppigy | i desperately need fat |
| 16:33.40 | Katty | cheese is not cheese unless it's cheese |
| 16:33.54 | Katty | eppigy: yes. you do. go eat another deluxe mcd breakfast |
| 16:34.11 | eppigy | It is 11:33am they wont sell me one :< |
| 16:34.11 | Katty | or perhaps a tin of pringles. |
| 16:34.16 | Katty | :< |
| 16:34.39 | jaytee | just stay away from the baked chips, they contain olestra and that causes anal leakage |
| 16:34.40 | Katty | now i want lasagna. |
| 16:34.49 | Katty | oh yes. good ole olestra. |
| 16:34.57 | Katty | commercial grade lubricant ingredient. |
| 16:35.04 | DocAwesome | o.O |
| 16:35.11 | Katty | it's true. wikipedia told me so. |
| 16:35.31 | jaytee | it's also found naturally in certain types of fish. pacific red snapper for one, any fish with a very pinkish meat |
| 16:35.35 | jaytee | pinkish orange |
| 16:35.46 | Katty | salmon's pink. |
| 16:35.57 | ManxPower-work | Joy. Our Digium analog card flaked out AGAIN. Happens about twice a month. |
| 16:36.10 | Katty | ManxPower-work: bummer. |
| 16:36.13 | Katty | ManxPower-work: i'd rma it |
| 16:36.39 | Katty | jaytee: so about your lasagna recipe... |
| 16:36.48 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:36.48 | ChannelZ | How does it flake out? |
| 16:36.49 | Katty | jaytee: do you actually layer it? or do you just mix it all together |
| 16:36.49 | jaytee | yeah? |
| 16:37.10 | jaytee | it wouldn't be real lasagna if I didn't layer it |
| 16:37.13 | ManxPower-work | Katty: won't do any good. |
| 16:37.23 | Katty | jaytee: mmmkay. |
| 16:37.46 | jaytee | and I don't use those lasagna noodles that you just bake without precooking because they end up tasting like cardboard |
| 16:37.52 | ManxPower-work | I think it's been replaced at least once already, Definatly swapped out for another card at least twice. I consider it just one of the oddities of the Digium analog cards. That's one of the reasons I tend not to use them. |
| 16:37.58 | jaytee | and they don't come in the right size |
| 16:38.17 | ManxPower-work | Katty: and you can't troubleshoot a problem that only happens randomly once or twice a month on a production PBX. |
| 16:38.56 | ChannelZ | Hmm. Thankfully I've never had mine do anything strange. |
| 16:39.12 | ManxPower-work | ChannelZ: seems to be related to number of calls between reboots |
| 16:39.21 | jaytee | I get the infrequent but random dropped calls over our PRI but I never see anything useful or notable in either the messages log or the debug log |
| 16:39.34 | Nugget | blame canada |
| 16:40.37 | Katty | ManxPower-work: ahh i see. |
| 16:40.47 | Katty | ManxPower-work: have you considered replacing it with a sangoma card? |
| 16:40.55 | ManxPower-work | ChannelZ: I've seen this issue for several years, across multiple generations of cards at multiple clients, on multiple servers. |
| 16:41.10 | ManxPower-work | Katty: Yes, we've been doing that with Digium cards that give us problems. |
| 16:41.15 | Katty | nods |
| 16:41.26 | Katty | http://www.recipezaar.com/Crock-Pot-Lasagna-21706 <- Yum-looking |
| 16:41.34 | thehar | ooooh |
| 16:41.40 | ManxPower-work | When we replaced the Digium T-1 card w/EC with the Sangoma all the echo issues went away. I think we are going to replace the analog card now. |
| 16:44.10 | ManxPower-work | We used to try troubleshooting issue. Ended up being cheaper just to replace the card with Sangoma. |
| 16:44.13 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 16:44.34 | Katty | must aquire juice and crackers at the store. |
| 16:44.39 | Katty | buhbye |
| 16:52.32 | *** join/#asterisk Tim_Toady (n=moi@77.49.49.66.dsl.dyn.forthnet.gr) |
| 16:53.47 | *** join/#asterisk stmaher (n=stephen@80.68.89.200) |
| 16:53.49 | stmaher | Hi everyone.. |
| 16:53.52 | stmaher | need emergency |
| 16:53.53 | stmaher | help |
| 16:54.10 | Deeewayne | O.o |
| 16:54.11 | stmaher | my driver for wctel12xp has gone missing after a machine reboot |
| 16:54.49 | stmaher | root@pbx01:/etc# modprobe wcte12xp |
| 16:54.49 | stmaher | Notice: Configuration file is /etc/zaptel.conf |
| 16:54.49 | stmaher | line 15: Cannot get number of tones for channel 1 |
| 16:55.32 | stmaher | root@pbx01:/etc# lsmod|grep wcte |
| 16:55.32 | stmaher | wcte12xp 39648 0 |
| 16:55.32 | stmaher | zaptel 195612 5 wcte12xp,ztdummy,zttranscode |
| 16:55.32 | stmaher | root@pbx01:/etc# |
| 16:55.50 | ManxPower-work | stmaher: it's there. |
| 16:55.51 | stmaher | zaptel_hardware shows nothing |
| 16:56.13 | stmaher | ok..can you please help me to figure out whats going wrong? |
| 16:56.21 | ManxPower-work | stmaher: not faimiliar with that utility. Does lspci show the card? |
| 16:56.35 | stmaher | 0000:03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) |
| 16:56.44 | dlynes | stmaher: zaptel_hardware? Is that a command in asterisk? Is that a command at the command line? |
| 16:56.52 | stmaher | commandline |
| 16:57.04 | ManxPower-work | stmaher: try power cycling the machine |
| 16:57.05 | Tim_Toady | stmaher ztcfg -fvvv |
| 16:57.40 | stmaher | root@pbx01:/etc# ztcfg -fvvv |
| 16:57.40 | stmaher | Notice: Configuration file is /etc/zaptel.conf |
| 16:57.40 | stmaher | line 15: Cannot get number of tones for channel 1 |
| 16:57.40 | stmaher | line 15: Cannot init tones for channel 1 |
| 16:57.45 | stmaher | 60 errors detected |
| 16:57.54 | Tim_Toady | hm |
| 16:58.10 | dlynes | stmaher: zaptel_hardware depends on an updated pci.ids file, that contains ids for your specific hardware |
| 16:58.12 | stmaher | the machine had 900 days uptime tho :-( |
| 16:58.22 | stmaher | ok |
| 16:58.26 | dlynes | stmaher: which you don't have |
| 16:58.42 | stmaher | dlynes ok? next move? |
| 16:59.11 | ManxPower-work | stmaher: power cycle the machine |
| 16:59.33 | stmaher | did that.. |
| 16:59.48 | ManxPower-work | stmaher: then your card might be bad. |
| 17:00.25 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 17:00.26 | ManxPower-work | stmaher: have you tried reinstalling zaptel? maybe something like libtonezone.0 was removed? |
| 17:00.36 | stmaher | might give that a go now |
| 17:00.38 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 17:00.38 | _abc_ | so how does one enable app_fax on 1.6 ? |
| 17:00.49 | _abc_ | where do i get something like that. google is not helpful |
| 17:00.51 | dlynes | grrr..stupid bug in firefox |
| 17:01.04 | _abc_ | dlynes: only one ? <duck> |
| 17:01.24 | _abc_ | fwiw asterisk 1.6.1 compiles cleanly in etch |
| 17:01.33 | dlynes | _abc_: install spandsp (http://www.soft-switch.org/), and then make distclean in your asterisk source tree, and then do configure again, and make install |
| 17:01.47 | ManxPower-work | _abc_: odd that /path/to/src/asterisk/doc didn't have any info on FAX. |
| 17:01.58 | _abc_ | thanks |
| 17:02.05 | dlynes | _abc_: assuming you don't have app_fax.so in your /usr/lib/asterisk/modules directory |
| 17:02.12 | _abc_ | ManxPower-work: i just grepped the tree i don't have the time to read ... |
| 17:02.24 | _abc_ | i just installed it now, it's still smoking |
| 17:02.24 | ManxPower-work | _abc_: then we don't have time to help you |
| 17:02.33 | _abc_ | ManxPower-work: thanks for letting me know that |
| 17:02.47 | _abc_ | dlynes: no i don't |
| 17:03.20 | *** join/#asterisk heliosj (n=jeff@i216-58-41-253.cybersurf.com) |
| 17:03.46 | dlynes | ManxPower-work: btw... _abc_ will be back here later on asking how he can switch codecs in the middle of a call, so that he can receive a fax |
| 17:03.49 | dlynes | blinks. |
| 17:04.25 | p3nguin | What the heck... I think people must listen to IVRs like they read IRC channel topics. I just monitored an incoming call, where the system played the "Thank you for calling [business name]. Please stay on the line blah blah blah..." then the representative answered, and the caller asked "is this [business_name]?" |
| 17:04.30 | _abc_ | is there a magic make argument that allows me to generate a tree for a binary deployment after this? excepting for --install-prefix ? |
| 17:04.42 | dlynes | _abc_: tar |
| 17:05.14 | _abc_ | dlynes: i mean a list of files to tar, dumped by make install presumably. i prefer pax or cpio usually |
| 17:05.16 | *** join/#asterisk davix (n=davix@212.199.161.41) |
| 17:05.27 | dlynes | p3nguin: i often do that, because the ivr's sound is so distorted that I didn't hear what it said |
| 17:05.33 | stmaher | I dont think the card is borked.. |
| 17:05.41 | stmaher | Im seeing two different versions of zaptel on the machine.. |
| 17:05.45 | stmaher | 1.2 and 1.4 |
| 17:05.53 | dlynes | _abc_: man find |
| 17:05.59 | p3nguin | _abc_: You want to make a package out of the software you just compiled? |
| 17:06.00 | _abc_ | also why do people default install sound files in gsm? i opted for ulaw |
| 17:06.06 | ManxPower-work | stmaher: you're screwed. |
| 17:06.13 | _abc_ | p3nguin: yes, so i can move it to another machine |
| 17:06.23 | dlynes | _abc_: because they can |
| 17:06.33 | stmaher | ManxPower-work Lovely.. just what I needed to hear |
| 17:06.36 | p3nguin | _abc_: What distro are you using? |
| 17:06.45 | dlynes | p3nguin: debian etch |
| 17:06.48 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 17:06.49 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 17:07.04 | p3nguin | Great! You can easily use checkinstall to take care of the hard work. |
| 17:07.17 | ManxPower-work | stmaher: the 1.4 install should have wiped out the 1.2 install |
| 17:07.32 | *** join/#asterisk kuku1 (n=ingo@c-67-175-3-155.hsd1.il.comcast.net) |
| 17:07.34 | stmaher | ManxPower-work Eh.. yeah you would think.. |
| 17:07.42 | stmaher | but like alot of other things when you enherit a network.. |
| 17:07.44 | stmaher | ................ |
| 17:08.23 | kuku1 | When I choose to record calls, the result is a wave, however, for the first half of the wav file, I hear one side of the conversation, and the second half of the wave I hear the other side of the conversation. Any ideas ? |
| 17:08.25 | p3nguin | _abc_: checkinstall will make a .deb out of your software, which you can send around to other debian machines. |
| 17:08.45 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 17:08.46 | p3nguin | _abc_: After all, you should always be using your package manager for everything, anyway. |
| 17:08.46 | _abc_ | p3nguin: thanks, that was about what i was looking for |
| 17:09.05 | _abc_ | p3nguin: package manager? what package manager? <g> |
| 17:09.11 | p3nguin | _abc_: Even if you compile your own software, you should still be making it into a package. |
| 17:09.13 | *** join/#asterisk ticoit (n=ticoit@190.241.180.89) |
| 17:09.56 | dlynes | kuku1: are you using mixmonitor(...)? |
| 17:10.06 | p3nguin | _abc_: "./configure && make && checkinstall" will probably get you pretty close. |
| 17:10.31 | p3nguin | Of course you can provide options to checkinstall so you don't have to answer a bunch of questions. |
| 17:10.50 | dlynes | kuku1: or are you just using monitor(...)? |
| 17:11.05 | stmaher | im reinstalling the drivers.. but it doesnt say antyhing about the hardware i have installed |
| 17:11.11 | stmaher | I think that the zaptel hardware you have on your system is: |
| 17:11.11 | stmaher | root@pbx01:~/voip/asterisk/svn/zaptel-1.4# |
| 17:14.01 | dlynes | Is there an updated doc on the pattern matching for dialplan extensions? |
| 17:14.16 | dlynes | Or is voip-info.org still the only source for that? |
| 17:15.41 | jaytee | the book is an excellent source for pattern matching. I've heard that some people actually even read it! |
| 17:16.33 | DocAwesome | lies!! |
| 17:16.39 | DocAwesome | jaytee: you shut your mouth |
| 17:16.47 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:16.48 | jaytee | :-) |
| 17:17.09 | kuku1 | dlynes: yes sir |
| 17:17.57 | dlynes | jaytee: smartass :) |
| 17:18.25 | kuku1 | dlynes: I'm using monitor, not mixmonitor |
| 17:18.57 | jaytee | dlynes, yep, I'm a smartass. I got this way from reading these things called books |
| 17:18.57 | heliosj | Books? |
| 17:19.16 | dlynes | kuku1: if you use mixmonitor, it'll blend both recordings together into a normal conversation |
| 17:19.19 | *** join/#asterisk dmz (n=dmz@118.sub-75-210-192.myvzw.com) |
| 17:19.25 | kuku1 | wow |
| 17:19.27 | kuku1 | I somehow missed that |
| 17:19.28 | dlynes | ~thebook |
| 17:19.29 | infobot | i heard thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 17:19.53 | heliosj | Is dat one of 'dem things with words and such? |
| 17:19.55 | dlynes | jaytee: i was just hoping there was an 'official' source from digium for it |
| 17:20.17 | dlynes | jaytee: much like asterisk-1.6.1.8/doc/*.tex |
| 17:20.43 | Katty | hi |
| 17:21.04 | dlynes | jaytee: you know? so that it would be updated for 1.6.1/1.6.2?...the book is for 1.4 |
| 17:21.42 | jaytee | dlynes, the book, while not stamped with the Digium Housekeeping Seal of Approval is given out and used in their training classes as a reference. |
| 17:22.10 | jaytee | dlynes, and pattern matching hasn't really changed between 1.4.x and 1.6.x as near as I can tell |
| 17:22.14 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 17:22.30 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 17:22.40 | dlynes | jaytee: yeah...i noticed in the book, 'Z' seems to be new....I don't remember anything other than N and X before |
| 17:23.08 | DocAwesome | Z has always been there |
| 17:23.12 | DocAwesome | just not used much |
| 17:24.44 | *** join/#asterisk lezland (n=root@mail.ivanics.hu) |
| 17:24.46 | lezland | hi |
| 17:25.09 | DocAwesome | hi |
| 17:25.15 | stmaher | reinstalling zaptel worked |
| 17:25.23 | stmaher | thanks for the moral support guys! |
| 17:25.30 | dlynes | DocAwesome: well...just found a use for it...was actually looking for something with that ability....probably why i found it |
| 17:25.35 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) [NETSPLIT VICTIM] |
| 17:25.36 | kuku1 | dlynes: mixmonitor exten => _NXXNXXXXXX/150,n,MixMonitor(wav,/asterisk/${CALLFILENAME}|m) doesnt record it |
| 17:25.42 | *** join/#asterisk ManxPOwer (n=EWieling@24.42.221.26) |
| 17:26.17 | paulc | Ah, another day another dollar.. except today it's database design, not much telecom related.. |
| 17:26.22 | DocAwesome | dlynes: you can also use something like _NXXNXX[256]XXX if you wanted only 2, 5, and 6 to match on the 7th digit |
| 17:26.36 | *** join/#asterisk Alagar (n=Administ@122.164.34.204) |
| 17:26.42 | raden_work | Is there any good billing software for asterisk to bill clients on a monthly basis ? |
| 17:27.33 | Kobaz | not really |
| 17:27.47 | Kobaz | there's some open source billing thing astbill or something |
| 17:27.57 | Kobaz | but it's kinda crappy |
| 17:28.20 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 17:28.31 | Kobaz | it's easier just to write a quick query on your cdr table, total up minutes by account code, and punch the number into quickbooks and off you go |
| 17:29.01 | lezland | I have an LDAP question: I'm trying to configure ldap module. Anonymous bind works fine, but I can't get it working with a username set. It will first tell me "Cannot connect to LDAP server" and then when I try to reload the res_config_ldap.so module it tells "Invalid DN syntax". have anyone experienced this DN syntax error message? how to debug it? |
| 17:29.59 | lezland | when I sniff the network traffic, it always shows some (uncrypted) traffic between asterisk and LDAP server |
| 17:30.09 | elliot98 | gives a big wave |
| 17:30.13 | elliot98 | hiya all |
| 17:31.22 | elliot98 | is there any problem with using Skype (not-asterisk Skype) and a SIP phone on the same computer? |
| 17:31.37 | *** join/#asterisk Heretic (n=fallen@dsl-246-111-150.telkomadsl.co.za) |
| 17:31.58 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 17:32.00 | [TK]D-Fender | elliot98: Why would they care? |
| 17:32.13 | [TK]D-Fender | elliot98: Unless they are fighting over your sound card resources... |
| 17:32.18 | kuku1 | I'm using mixmonitor but instead of a wav file I get a raw file. |
| 17:32.21 | elliot98 | from a technical point of view |
| 17:32.37 | [TK]D-Fender | elliot98: technially.... it doesn't matter. |
| 17:33.06 | elliot98 | I'd assume SIP and Skype use different ports, and Windows should work out the sound issues |
| 17:33.09 | elliot98 | woops! |
| 17:33.20 | elliot98 | s/use/use\ different |
| 17:33.27 | elliot98 | not thinking today |
| 17:33.50 | [TK]D-Fender | elliot98: Completely different |
| 17:34.25 | elliot98 | see, I am trying to find out why sometimes a call is answered automatically on an X-lite softphone |
| 17:35.10 | elliot98 | doesn't always happen, but there a few computers set up in a queue with softphones and sometimes one of the softphones just answers |
| 17:35.37 | elliot98 | so I was looking at some networking conflich possibilities |
| 17:35.51 | Kobaz | extra sip headers are being set... like alert-info (which is the usual header for autoanswer control) |
| 17:35.57 | *** join/#asterisk spiegel (n=spike@ip-166-187.interbild.net) |
| 17:36.13 | Kobaz | paste your sip debug for a call that auto answers |
| 17:36.16 | [TK]D-Fender | Kobaz: Which... X-Lite doesn't support |
| 17:36.32 | Kobaz | [TK]D-Fender: hmm, mayber it's the pro version that supports it... |
| 17:37.15 | elliot98 | but it doesn't always happen |
| 17:37.23 | Kobaz | sip debug |
| 17:37.29 | elliot98 | I am thinking more that it is a queue issue |
| 17:37.33 | Kobaz | SIP DEBUG |
| 17:37.35 | elliot98 | something buggy in the queue |
| 17:37.42 | elliot98 | gotcha |
| 17:37.48 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 17:39.07 | elliot98 | how do I debug specific devices? |
| 17:39.26 | _abc_ | erm p3nguin there is no checkinstall on my system |
| 17:39.30 | Kobaz | sip set debug ip |
| 17:39.37 | Kobaz | type: sip set debug <tab> |
| 17:39.41 | Kobaz | to see all the debug options |
| 17:40.39 | p3nguin | _abc_: I guess you had better install it, then. |
| 17:41.23 | _abc_ | looks like it |
| 17:41.52 | p3nguin | It can't be too terribly difficult to "apt-get install checkinstall" |
| 17:42.06 | _abc_ | p3nguin: you don't want to know |
| 17:42.47 | diatonic | I'm seeing this bug in 1.4.26: https://issues.asterisk.org/view.php?id=12497 - Shouldn't it be fixed in 1.4.26? New to asterisk, not sure if I should file new bug report |
| 17:42.51 | elliot98 | how much bandwidth does skype take up? |
| 17:42.58 | p3nguin | all of it |
| 17:43.14 | elliot98 | all of it? |
| 17:43.20 | Katty | someone take these crackers away from eme before i eat them all |
| 17:43.24 | *** join/#asterisk ruied (n=ruied@89.214.64.233) |
| 17:43.37 | elliot98 | give 'em to your coworker |
| 17:43.43 | elliot98 | tell her you got them from the internet |
| 17:43.45 | Katty | hells no. |
| 17:44.50 | elliot98 | squirrels? |
| 17:45.03 | Katty | >.< |
| 17:45.06 | Katty | i'm not done with the crackers! |
| 17:45.08 | Katty | just..for now |
| 17:45.25 | ChannelZ | Skype like anything depends on the codec |
| 17:46.05 | *** join/#asterisk MaliutaLap (n=biteme@204.239.250.1) |
| 17:47.18 | elliot98 | yes, but I've been reading that they're this peer-to-peer messaging thing |
| 17:47.44 | elliot98 | I imagine peer-to-peer can be a big load on VOIP upload |
| 17:48.06 | raden_work | is there a directory of area codes in the continental USA ? |
| 17:48.08 | ChannelZ | Yes if you become a supernode you can be passing other data |
| 17:48.59 | ChannelZ | but I don't believe you're actually streaming any voice traffic |
| 17:49.08 | elliot98 | aha...so here I have like 5+ computers with Skype, |
| 17:49.15 | elliot98 | one may very well becoming a supernode |
| 17:49.23 | elliot98 | and messing with the SIP devices |
| 17:49.43 | _abc_ | dlynes: you have been very helpful |
| 17:49.57 | ChannelZ | potentially. Google for it, there are some registry hacks for WIndows anyway where you can tell them not to become supernodes |
| 17:49.57 | _abc_ | thanks, i have done everything i needed to do today |
| 17:50.12 | *** part/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 17:50.42 | drmessano | Even though the traffic sent to supernodes is negligible, some institutions are interested in preventing users on their network from becoming supernodes and, thereby, answering directory enquiries for other users. |
| 17:50.49 | drmessano | From their FAQ |
| 17:50.56 | drmessano | Its not like you're seeding a torrent |
| 17:51.09 | ChannelZ | Me too. I haven't figured out how to make SFA not become a supernode |
| 17:51.37 | drmessano | Also says if you're NAT'ed, you wont |
| 17:51.49 | ChannelZ | (actually I'm not sure if it is, it might just be talking to others, outgoing...) |
| 17:51.51 | drmessano | ChannelZ: SFA isn't a traditional Skype client |
| 17:51.51 | romain | does someone knows why there isn't any lock on any mutex in the meetme's conf_run main loop? |
| 17:52.03 | ChannelZ | yah I just thought of that |
| 17:52.10 | *** join/#asterisk MaliutaLap (n=biteme@204.239.250.1) |
| 17:52.16 | Naikrovek | polycom people; anyone know how to configure the little messages button that comes with the phones |
| 17:52.29 | Naikrovek | you replace a line button with the messages button |
| 17:52.31 | Naikrovek | and you can set it up to auto dial a number fo ryou |
| 17:52.32 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:52.32 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 17:52.32 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 17:52.32 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 17:52.32 | *** join/#asterisk StevenR (n=foo@wan1.sghs.org.uk) |
| 17:52.32 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 17:52.32 | *** join/#asterisk yoruk_ (n=yoruk@host164-182-dynamic.52-82-r.retail.telecomitalia.it) |
| 17:52.32 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 17:52.34 | elliot98 | SFA? |
| 17:52.34 | Naikrovek | or something |
| 17:52.56 | drmessano | SkypeForAsterisk |
| 17:52.59 | ChannelZ | I always get connections to wierd dynamic hosts on my * box but now that I think of it they are outgoing connections |
| 17:54.09 | [TK]D-Fender | Naikrovek: Restrict your reg to 1 linekey, and make a directory entry from vm. the End. |
| 17:55.37 | elliot98 | I'm not using SFA |
| 17:55.43 | elliot98 | just plain old skype |
| 17:56.27 | Naikrovek | [TK]D-Fender: okay |
| 17:56.41 | elliot98 | from what I am reading, Skype likes to become bandwidth happy |
| 17:56.47 | drmessano | elliot98: and your issue was addressed based on that.. shall we go over ti again? |
| 17:57.08 | drmessano | According to their FAQ, thats not true |
| 17:57.16 | drmessano | I even pasted from it |
| 17:57.44 | drmessano | We also mentioned the registry hack, which will prove further that your b/w issues arent due to becoming a supernode |
| 17:57.56 | drmessano | But this is way off topic at this point |
| 17:58.26 | elliot98 | I'll check out, um networking channels |
| 18:01.30 | jdnwest | lol, Bandwidth is cheap (Stateside). |
| 18:01.47 | MaliutaLap | cadnadia, oh canadia ... something, something, something |
| 18:02.35 | ChannelZ | cheap in quality sure |
| 18:03.03 | jdnwest | Ok, i'm going to have to ask, as long as your pinging under 80, why does it matter? |
| 18:03.08 | jdnwest | And it Stays UP. |
| 18:03.19 | drmessano | Stateside? In the sense the US is like 15th in the world with regard to available bandwidth and pricing? |
| 18:04.09 | drmessano | jdnwest: if you're pinging 80, your line stays up, and all your calls sound like shit, people dont care about the first twp |
| 18:04.10 | jdnwest | Drmessano, If i'm buying by the gigE port Its between 3 and a 1.5 a meg. |
| 18:04.11 | drmessano | two* |
| 18:05.28 | jdnwest | As long as you've got uptime, latency, and jitter down, what is this magical "Quality" everyone talks about, because I don't understand the difference in quality between L3 and TWtelecom even though the pricing is miles apart. |
| 18:06.35 | drmessano | This is gonna be another one of those moving target conversations I have no time for.. got a radiothon to set up, bbl |
| 18:06.48 | jdnwest | lol |
| 18:08.26 | jdnwest | Has anyone had a tier 1 or 2 carrier that provided bandwidth that was "unusable" for VOIP? |
| 18:14.05 | *** join/#asterisk rdahlin_1 (n=rdahlin_@78-73-17-198-no168.tbcn.telia.com) |
| 18:17.51 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:19.36 | *** join/#asterisk the_limit (n=the_limi@75-150-44-61-Oregon.hfc.comcastbusiness.net) |
| 18:22.03 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 18:22.10 | Katty | brr. |
| 18:22.16 | _abc_ | i am confused. whats the command to reload a dialplan in 1.6 ? |
| 18:22.37 | _abc_ | Katty: why is it you do a brr from time to time? |
| 18:22.43 | Katty | because it is brr |
| 18:22.48 | _abc_ | cold? |
| 18:22.48 | eppigy | brr |
| 18:22.49 | eppigy | brrrr |
| 18:22.51 | Katty | yes. |
| 18:22.51 | PoeticIntensity | probably because it's cold. |
| 18:22.52 | Katty | brr. |
| 18:23.04 | _abc_ | sounds like a horse |
| 18:23.12 | _abc_ | again: i am confused. whats the command to reload a dialplan in 1.6 ? |
| 18:23.13 | Katty | that's neh |
| 18:23.26 | Katty | possibly nah |
| 18:23.27 | Katty | ney |
| 18:23.28 | Katty | nay |
| 18:23.30 | Katty | one of those. |
| 18:23.36 | _abc_ | i used to do reload dialplan or reload extensions now its 'deprecated' how helpful |
| 18:23.56 | MaliutaLap | waves to Katty |
| 18:24.00 | ChannelZ | module reload pbx_config |
| 18:24.08 | jaytee | _abc_, did you try dialplan reload? |
| 18:24.11 | MaliutaLap | Katty: been in .ca for a few hours ... no squirrels |
| 18:24.27 | Katty | MaliutaLap: squirrels are mostly out in the wee hours of the morning |
| 18:25.17 | MaliutaLap | been here since 07:30 local ... that _is_ the wee hours ;) It's before 12:00 |
| 18:25.38 | MaliutaLap | Katty: do the squirrels live in the airports anyhow? |
| 18:25.50 | Katty | ^_- |
| 18:25.53 | Katty | what do you think |
| 18:26.01 | Katty | if you were a squirrel, would you want to live in an airport? |
| 18:26.08 | MaliutaLap | I think they're squirrels and they might |
| 18:26.21 | Katty | maybe you should put out pecans. |
| 18:26.39 | MaliutaLap | aren't they the ones living under the ground and stealing vegetables? |
| 18:26.44 | MaliutaLap | plays dumb |
| 18:26.51 | Katty | facepalms |
| 18:27.05 | MaliutaLap | :) |
| 18:27.10 | MaliutaLap | it was a long flight |
| 18:27.13 | Katty | hugs MaliutaLap |
| 18:27.20 | MaliutaLap | yay! hugs! |
| 18:27.28 | eppigy | man i would be a plane hijacking squirrel |
| 18:27.40 | Katty | eppigy: no passport required |
| 18:27.40 | MaliutaLap | * let me talk to my mummy! |
| 18:29.21 | MaliutaLap | on descent I told the girl next to me that they sound proofed the cockpit for those moments ... so you can't hear the alarms as the plane falls from the air :) |
| 18:30.46 | _abc_ | how do i set a nonstandard sip port for a friend client in sip.conf? |
| 18:31.41 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 18:32.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:32.59 | p3nguin | _abc_: Is your * server listening on that nonstandard port? |
| 18:33.31 | MaliutaLap | or are you at least NATing to it? |
| 18:34.10 | _abc_ | p3nguin: i have the server on standard port and a local linphone on 5065 |
| 18:34.21 | _abc_ | i edited its config for that |
| 18:35.08 | p3nguin | Just because you change the SIP port on your phone doesn't necessarily mean that it's not going to talk to * on the regular SIP port. |
| 18:35.25 | _abc_ | so do i have to set the phone's port number in its sip.conf entry or does * take it from the invite ? |
| 18:35.27 | ManxPOwer | remember all packets have 2 ports, a source port and a destination port. |
| 18:35.40 | *** part/#asterisk ManxPOwer (n=EWieling@24.42.221.26) |
| 18:36.26 | _abc_ | sure the question was, do i have to set it or does * take it from the registration and attempts to ring at that non standard sip port? |
| 18:36.53 | _abc_ | my phone registers * likes it it is in sip show peers and i can't call |
| 18:38.37 | *** join/#asterisk dmz (n=dmz@eth0.dhcp1.sfo2.servepath.net) |
| 18:39.29 | _abc_ | ok sip show peers shows the nonstandard port ok |
| 18:39.47 | elliot98 | I am getting a huge amount of console logins: |
| 18:39.48 | elliot98 | http://pastebin.com/d1498a8cf |
| 18:40.00 | elliot98 | they all are coming from the same process though |
| 18:40.03 | elliot98 | how is this happening? |
| 18:40.05 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:40.06 | _abc_ | strangely tcpdump shows that the phone connects to * via sip and then nothing happens |
| 18:41.14 | elliot98 | that process is the main Asterisk process |
| 18:43.31 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 18:47.37 | elliot98 | is this a bug or what? |
| 18:49.11 | [TK]D-Fender | elliot98: Stop freaking out and show us a problem :) |
| 18:49.28 | Katty | shows [TK]D-Fender a cut on her thumb |
| 18:49.31 | Katty | :< |
| 18:49.46 | [TK]D-Fender | Katty: Bet you mine was worse... |
| 18:53.06 | Katty | :< |
| 18:53.13 | drfreeze | Anyone know how to set an IP address on a Polycom 331 phone? |
| 18:53.29 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 18:53.29 | drfreeze | I have one phone that I can't seem to figure out how to set it |
| 18:53.41 | drfreeze | All I can type in is 192*168*0*50* |
| 18:53.43 | [TK]D-Fender | drfreeze: Boot phone. Enter "Setup". Set IP address |
| 18:53.53 | drfreeze | How do I enter '.' |
| 18:54.04 | Katty | ##? |
| 18:54.05 | [TK]D-Fender | drfreeze: "*" |
| 18:54.10 | ManxPower-work | drfreeze: on Polycoms * = . |
| 18:54.11 | elliot98 | I posted the netstat |
| 18:54.23 | drfreeze | hrm |
| 18:54.24 | drfreeze | ok |
| 18:54.29 | Katty | drfreeze: i seem to recall as imliar problem on the 330s |
| 18:54.32 | [TK]D-Fender | elliot98: Forget netstat. Show us a broken call |
| 18:54.34 | elliot98 | and now I am getting "too many connections errors" when I asterisk -r |
| 18:54.42 | Katty | drfreeze: let me go find one |
| 18:54.56 | [TK]D-Fender | elliot98: Stop * completely. Restart. |
| 18:55.06 | *** join/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 18:55.09 | elliot98 | I did that...everything is ok now |
| 18:55.17 | elliot98 | but this is not an uncommon occurence |
| 18:55.34 | ManxPower-work | elliot98: Yes, it is an uncommon occurance. |
| 18:55.56 | elliot98 | well, not for me |
| 18:56.04 | elliot98 | happens say, once every two days |
| 18:56.06 | ManxPower-work | maybe, but it is uncommon for everyone else. |
| 18:56.34 | elliot98 | what could be the cause of asterisk suddenly duplicating its process? |
| 18:57.04 | ManxPower-work | elliot98: the question you should be asking is "what is running so many copies of "asterisk -r" |
| 18:57.37 | Katty | drfreeze: this one makes the * show up like a . <- maybe firmware change? |
| 18:58.13 | elliot98 | but the netstat states it's all the same process |
| 18:58.23 | elliot98 | for example, if I do an asterisk -r |
| 18:58.27 | elliot98 | a different process shows up |
| 18:58.28 | [TK]D-Fender | elliot98: What version are you running? |
| 18:58.32 | elliot98 | 1.4 |
| 18:58.34 | [TK]D-Fender | elliot98: Installed how? |
| 18:58.41 | [TK]D-Fender | elliot98: what version EXACTLY? |
| 18:58.42 | elliot98 | from source |
| 18:58.47 | elliot98 | oh, right, 1.4.18 |
| 18:58.54 | [TK]D-Fender | elliot98: ANCIENT. go upgrade |
| 18:59.16 | elliot98 | yeah...haven't done that in a while |
| 18:59.37 | elliot98 | from your experience, you think it's a version thing? |
| 18:59.38 | [TK]D-Fender | elliot98: Nver ask to solve problems in old versions... if they aren't current, then it HAD problems :) |
| 19:00.02 | _abc_ | this is really strange. i am registered with a sip client, the extensions.conf is standard, and the phone tells me it can't connect on any of the demo numbers, i.e. 500, 600 etc |
| 19:00.08 | _abc_ | what could cause this? |
| 19:00.19 | [TK]D-Fender | _abc_: No such thing as "standard" |
| 19:00.34 | [TK]D-Fender | _abc_: And you aren't looking at the call |
| 19:00.35 | _abc_ | [TK]D-Fender: well straight dist |
| 19:00.41 | [TK]D-Fender | _abc_: Meaningless |
| 19:00.53 | _abc_ | i did the usual things, sip show peers and core show channels |
| 19:00.58 | _abc_ | one peer, no channels |
| 19:01.00 | Katty | eppigy: do you have any neosporin |
| 19:01.04 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:01.05 | ManxPower-work | _abc_: The *.sample config files are NOT intended to be a working system. |
| 19:01.10 | [TK]D-Fender | _abc_: Go look at SIP DEBUG for the incoming call and look where it is looking to match it |
| 19:01.22 | _abc_ | hmm |
| 19:02.12 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 19:03.15 | *** join/#asterisk ecrist (n=ecrist@pdpc/supporter/professional/ecrist) |
| 19:03.25 | ecrist | anyone in here familiar with polycom bitmap configuration? |
| 19:03.57 | diatonic | ecrist: I've gotten custom bitmaps on IP 501s, 550s and 650s |
| 19:04.29 | *** join/#asterisk MaliutaLap (n=biteme@204.239.250.1) |
| 19:04.34 | ecrist | diatonic: I'm trying to get a custom background bitmap on our 330s |
| 19:04.47 | ecrist | when the phone boots, the logo shows up for a split second then goes away |
| 19:05.40 | diatonic | ecrist: Have you looked at this? http://www.voip-info.org/wiki/view/Polycom+Idle+Images |
| 19:05.42 | ecrist | it appears to want to take up about the bottom 2/3 of the display |
| 19:06.20 | ecrist | no, quick glance it looks a lot like the polycom admin guide, but I'll read it quick |
| 19:06.39 | diatonic | ecrist: Looks like a 330 needs a B&W image, 102x23 pixels |
| 19:06.44 | drfreeze | Silly phone |
| 19:06.59 | ecrist | I have that |
| 19:07.06 | drfreeze | It converts the 192*168*0*50 to 192.168.0.50 for the SNTP server, but not the FTP server |
| 19:07.12 | drfreeze | and it can't find the boot server |
| 19:07.29 | eppigy | Katty: i have some |
| 19:08.07 | drfreeze | Trying to set the ftp server from the web interface, but I don't see where to do that |
| 19:09.01 | _abc_ | [TK]D-Fender: it was the realm. i had to set it to localhost for this (local) host |
| 19:09.07 | _abc_ | thanks for the tip |
| 19:09.41 | drfreeze | Anyone have some pointers on how to set the IP address for the FTP |
| 19:09.50 | MaliutaLap | vim |
| 19:10.36 | ecrist | drfreeze: what phone? |
| 19:10.42 | diatonic_ | drfreeze: DHCP Reservation? |
| 19:11.03 | ManxPower-work | drfreeze: why not set tftp-server in your DHCP server setup? |
| 19:12.10 | diatonic_ | drfreeze: If you have more than a few Polycoms you should be setting all that stuff in DHCP for easy provisioning |
| 19:12.27 | diatonic_ | is really going to lunch now |
| 19:12.35 | *** join/#asterisk wierdo (n=jimmy@89.252.206.114) |
| 19:13.26 | drfreeze | ecrist: Polycom 331 |
| 19:13.49 | ecrist | option 61, also named tftp-server as ManxPower-work said |
| 19:14.06 | drfreeze | ManxPower-work: it's a winders network running the dhcp |
| 19:14.12 | *** join/#asterisk path (i=path@server1.bshellz.net) |
| 19:14.15 | drfreeze | phones and computers are on the same network - existing office |
| 19:14.22 | p3nguin | option 61? |
| 19:14.42 | bcrisp | anyone here used Queue-Tip queue monitoring software? |
| 19:14.57 | ecrist | drfreeze: I'll post my working dhcp server config for you, hang on |
| 19:15.02 | p3nguin | You'll probably want to use 66 and/or 150 for setting the tftp address. |
| 19:15.46 | drfreeze | ecrist: cool |
| 19:16.16 | ecrist | p3nguin: 66, my mistake |
| 19:16.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.12 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 19:18.32 | ecrist | drfreeze: http://pastebin.com/m709271c0 |
| 19:18.57 | ecrist | phones and all computers are on the same lan (many computers are plugged into the phones, using the built-in switch) |
| 19:18.59 | dlynes | Anyone have any idea what '[Dec 9 11:17:40] WARNING[8361]: chan_sip.c:16493 handle_response_invite: just did sched_add waitid(15840894) for sip_reinvite_retry for dialog 000067dd-e1feeade6763100097640080f03ca102@xx.xx.xx.xx in handle_response_invite means? |
| 19:19.22 | *** join/#asterisk rbd (n=rbd@rrcs-98-101-33-14.midsouth.biz.rr.com) |
| 19:19.25 | ecrist | using the vendor-class-identifier, polycom phones are assigned an IP within a given range (QoS on firewall is handled this way) |
| 19:19.37 | ecrist | all other systems are delegated otherwise |
| 19:21.17 | *** join/#asterisk voipmonk (n=voipmonk@69.172.100.53) |
| 19:26.26 | *** join/#asterisk Ta^3 (n=tacvbo@189.146.170.87) |
| 19:33.34 | _abc_ | i am unable to find concise documentation on things like the complete argument list for MusicOnHold() and the like. where should i look? in the asterisk book? |
| 19:34.06 | *** part/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 19:34.10 | ManxPower-work | _abc_: All official app docs can be seen with "core show application theappyouwant" |
| 19:34.12 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 19:34.29 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.233.45.dsl.dyn.forthnet.gr) |
| 19:34.31 | ManxPower-work | you can see a list of installed application with "core show applications |
| 19:34.31 | Katty | hi crispy |
| 19:34.35 | *** join/#asterisk wierdo (n=jimmy@89.252.206.114) |
| 19:34.43 | _abc_ | ManxPower-work: thanks |
| 19:36.50 | [TK]D-Fender | _abc_: http://pastebin.ca/1708643 |
| 19:36.56 | _abc_ | this is what i was looking for: http://www.asterisk.org/node/48581 |
| 19:37.16 | _abc_ | hehe [TK]D-Fender |
| 19:38.49 | Nugget | heh |
| 19:39.47 | [TK]D-Fender | ... |
| 19:39.48 | [TK]D-Fender | telnet |
| 19:39.50 | bcrisp | is there a different channel i should use for discussing AMI ? |
| 19:40.01 | [TK]D-Fender | Nugget: you're no fun... you know that? :p |
| 19:40.06 | [TK]D-Fender | bcrisp: Nope |
| 19:40.25 | Nugget | pfft |
| 19:41.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:43.25 | _abc_ | any ideas why MusicOnHold() (<- notice no arguments) stops playing after the 1st song and does not loop |
| 19:44.18 | p3nguin | How are you implementing it? |
| 19:45.57 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 19:46.58 | _abc_ | p3nguin: files, native codec format (ulaw in my case) |
| 19:47.46 | p3nguin | How are you running the MusicOnHold() command? |
| 19:48.22 | _abc_ | exten => 700,1,MusicOnHold() |
| 19:48.35 | _abc_ | and i was wrong it does not play to the end |
| 19:48.40 | _abc_ | it just cuts off |
| 19:48.44 | _abc_ | i did not time it |
| 19:48.48 | _abc_ | the line is not hung up |
| 19:49.17 | _abc_ | in the console i get this: -- Started music on hold, class 'default', on SIP/phoneid-00000008 |
| 19:49.29 | p3nguin | Change it away from 700, first, since that is typically used for parking. Then second, put an Answer() before MusicOnHold(). |
| 19:49.45 | _abc_ | i turned sip debug on now |
| 19:50.05 | _abc_ | p3nguin: whats the role of Answer ? |
| 19:50.12 | p3nguin | to answer the channel |
| 19:50.35 | _abc_ | 700 is not used in my setup, no 700 exten anywhere |
| 19:50.40 | p3nguin | okay |
| 19:51.01 | p3nguin | Answer before MoH, and your problem will go away. |
| 19:52.34 | _abc_ | i put in Answer as you said, still on 700, killed after 10 sec |
| 19:52.39 | _abc_ | faster then before ;) |
| 19:52.44 | _abc_ | moving off of 700 ... |
| 19:52.59 | p3nguin | Maybe you're doing it wrong. |
| 19:53.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:53.49 | p3nguin | exten => 700,1,Answer() exten => 700,n,MusicOnHold() |
| 19:54.12 | Katty | MusicOnHold(foogroup) |
| 19:54.25 | _abc_ | Katty: the default should work fine |
| 19:54.27 | p3nguin | I have an exten just like yours that I use to monitor my MoH stream. I can listen to hold music for HOURS if I want to. |
| 19:54.31 | _abc_ | there is only one |
| 19:54.45 | Katty | foogroup. kthx. |
| 19:54.53 | _abc_ | p3nguin: i know i used to use that before on an older asterisk and never had a problem |
| 19:55.15 | p3nguin | If I don't Answer() first, it exits after a short amount of time. |
| 19:55.22 | _abc_ | is there something in asterisk that downs moh threads if system load is too high? |
| 19:55.35 | _abc_ | i answer it now alright and i moved to 800 |
| 19:56.03 | p3nguin | Okay, so you call 800, it answers the channel and then plays music. |
| 19:56.31 | _abc_ | sure it has been playing for 2 minutes now. |
| 19:56.41 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:56.43 | _abc_ | but the key was to move off of 700 not Answer() |
| 19:57.02 | _abc_ | first i added Answer() still on 700 and it hung up on me |
| 19:57.06 | _abc_ | the thread |
| 19:57.14 | p3nguin | I guess you missed the part where I said if I didn't answer my channel first that it would exit the music. |
| 19:57.23 | _abc_ | then i moved it to 800 with Answer and the jukebox seems to work now ;) |
| 19:58.03 | p3nguin | I'm curious about something, now. I'm wondering if I don't answer it first, will core show channels show an up channel or a ringing channel? |
| 19:58.17 | _abc_ | try it |
| 19:58.19 | p3nguin | I should test. |
| 19:58.58 | _abc_ | i like the new moh music. the old one was great but this is also good |
| 19:59.01 | p3nguin | Ring MusicOnHold() |
| 19:59.16 | p3nguin | How long will * leave a channel in the Ring state? |
| 19:59.21 | Nugget | replaces _abc_'s hold music with rick astley |
| 19:59.22 | _abc_ | p3nguin: does that work? Ring(MusicOnHold()) ? |
| 19:59.57 | p3nguin | uh, no |
| 19:59.59 | _abc_ | p3nguin: i think that its a SIP default setting, the timeout |
| 20:00.19 | _abc_ | and its different for different channels |
| 20:00.20 | p3nguin | core show channels: State Ring, Application(data) MusicOnHold() |
| 20:00.44 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 20:00.49 | p3nguin | So once the ring state timeout is reached, MoH is going to exit. |
| 20:00.59 | p3nguin | And that was your complaint. |
| 20:01.33 | p3nguin | But if you Answer() the channel first, then it goes into State Up. |
| 20:01.50 | p3nguin | timedout |
| 20:01.56 | [TK]D-Fender | Alrighty... they're letting out ealry due to our first snow storm... BBL |
| 20:02.16 | p3nguin | so roughly 2 minutes to timeout. |
| 20:04.10 | _abc_ | ok, the music thing works great |
| 20:04.23 | _abc_ | loops and all that, 11 minutes and going |
| 20:04.47 | _abc_ | is there a goodish security faq/intro for the network side of asterisk? i want to put it in a chroot eventually |
| 20:04.51 | p3nguin | Now you can transfer annoying callers to 800. |
| 20:04.53 | _abc_ | or in a virtual host |
| 20:05.56 | _abc_ | p3nguin: i can use it to test whether the pbx is up and not frozen among others |
| 20:06.16 | _abc_ | and i have a mancini music collection i will put up ;) |
| 20:06.34 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:06.49 | p3nguin | echo test would be fine for testing that. |
| 20:07.05 | _abc_ | p3nguin: i don't like to talk to myself |
| 20:07.11 | p3nguin | :) |
| 20:07.12 | _abc_ | it feels odd |
| 20:07.35 | _abc_ | so what would be a goodish security faq? |
| 20:07.38 | _abc_ | for asterisk |
| 20:07.56 | ManxPower-work | _abc_: I doubt one exists. |
| 20:07.58 | p3nguin | http://blogs.digium.com/2009/03/28/sip-security/ |
| 20:08.03 | ManxPower-work | Or maybe one does. |
| 20:08.03 | p3nguin | That's a good start. |
| 20:09.10 | dlynes | _abc_: You can try: http://www.google.ca/search?hl=en&q=site%3Asecurityfocus.com+asterisk&btnG=Search&meta=&aq=f&oq= |
| 20:09.37 | _abc_ | heh dlynes thanks |
| 20:09.43 | drfreeze | Ok, got another problem with a Polycom 331 |
| 20:09.55 | dlynes | _abc_: That's a third party independent security site that handles security flaws in various softwares |
| 20:09.56 | _abc_ | that's rather the post mortem stuff. what about prevention? |
| 20:10.14 | _abc_ | dlynes: i know i have been subscribed to it for years |
| 20:10.19 | dlynes | ah |
| 20:10.19 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:10.25 | drfreeze | This phone was working and I switched network ports, and it no longer gets an IP address |
| 20:10.35 | _abc_ | but they tend to point out problems after they exist not before |
| 20:10.45 | dlynes | drfreeze: maybe it's on a different vlan now? or hard-codec for a different speed or duplex? |
| 20:10.47 | _abc_ | so i was looking for the securing docs |
| 20:11.15 | dlynes | _abc_: put it into a VPE or VPS, or chroot it |
| 20:11.16 | p3nguin | _abc_: Alpha-numeric peer names, alpha-numeric secrets with not less than 12 characters, if you know the IP addresses of your peers/users, specify them and/or create ACLs... |
| 20:11.30 | dlynes | _abc_: if you're using iax, you can additionally use keys |
| 20:11.45 | _abc_ | hmm iax is an option |
| 20:12.07 | ManxPower-work | _abc_: 1) secure your operating system 2) follow that blog post. |
| 20:12.11 | _abc_ | does the iax protocol use mutiple firewall piercing ports or just one? |
| 20:12.18 | p3nguin | 1 |
| 20:12.26 | dlynes | _abc_: 4569 only, unless you specify a different one |
| 20:12.29 | *** join/#asterisk StevenR_ (n=foo@wan1.sghs.org.uk) |
| 20:12.30 | dlynes | _abc_: udp |
| 20:12.38 | _abc_ | so its superior in this respect to sip which is all over the place |
| 20:12.40 | _abc_ | good |
| 20:12.53 | dlynes | _abc_: i abhor sip for firewalls, too...but it is what it is |
| 20:13.07 | _abc_ | sip should stand for 'sieve in protection' |
| 20:13.08 | dlynes | _abc_: and if you want to interface with third party hardware, you don't have a choice |
| 20:13.09 | ManxPower-work | IAX2 has it's own issues |
| 20:13.10 | p3nguin | SIP is just one port, too... but the audio goes into a specified range. It's not really all over the place. |
| 20:13.27 | vk2dgy | does anyone know if it's possible to issue multiple commands at a time via asterisk -rx ? |
| 20:13.28 | vk2dgy | eg, I want to do: "sip show peers" and "show channels" (and possibly others) without |
| 20:13.29 | vk2dgy | having to make a series of calls to asterisk -rx |
| 20:13.42 | *** join/#asterisk ruied (n=ruied@89.214.64.233) |
| 20:13.43 | _abc_ | p3nguin: okay, yes, but it tries to be peer to peer and other things which are no-nos in nat systems |
| 20:13.57 | p3nguin | Not really. canreinvite=no |
| 20:14.04 | dlynes | _abc_: for remote end points where it doesn't make sense to have a pc, we use sip, we use sip over a private lan for our upstream, and between all of our satellites, we use iax2 |
| 20:14.10 | *** join/#asterisk DigitalFlux (n=DigitalF@unaffiliated/digitalflux) |
| 20:14.18 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 20:14.33 | _abc_ | dlynes: ok, that's because iax2 is also a trunking protocol which sip isn't |
| 20:14.38 | _abc_ | right? |
| 20:14.55 | dlynes | _abc_: no...just makes it easier, and I don't have to deal with firewalls |
| 20:15.04 | dlynes | _abc_: and iax2 gives you more debugging info |
| 20:15.13 | _abc_ | i see. but yes, those two concepts are related |
| 20:15.27 | dlynes | _abc_: and iax2 can also share dialplans between machines, for whatever that's worth |
| 20:15.45 | dlynes | _abc_: i see it as more of a security risk than anything, but that's my opinion |
| 20:16.00 | dlynes | _abc_: you can also further secure it by only allowing connections from your vpn networks |
| 20:17.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.35 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 20:18.13 | _abc_ | yeah but that's becoming complex quickly |
| 20:18.36 | dlynes | _abc_: you asked...I told you |
| 20:18.37 | _abc_ | a few years ago i wrote auto provisioning scripts for nortel phones, i'll thaw that and see if it still works |
| 20:18.55 | _abc_ | is someone here using asterisk with cisco phones by chance? with skinny? |
| 20:18.59 | _abc_ | or sccp? |
| 20:19.07 | dlynes | _abc_: lots.... |
| 20:19.11 | _abc_ | anything to look out for? |
| 20:19.11 | hardwire | sigh.. I have asterisk 1.6.2 registering to iax peers on an asterisk 1.2 system |
| 20:19.13 | dlynes | points at Qwell. |
| 20:19.30 | dlynes | comforts hardwire. |
| 20:19.31 | _abc_ | do the 79xx's work well with asterisk ? |
| 20:19.31 | hardwire | calls from 1.6.2 to 1.2 work fine. |
| 20:19.32 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 20:19.40 | hardwire | the other direction doesn't match up the peers correctly |
| 20:19.43 | dlynes | hardwire: wow...I'm surprised |
| 20:19.43 | hardwire | it thinks it's an iax guest |
| 20:19.55 | Qwell | what? |
| 20:20.01 | dlynes | hardwire: I had issues in the other direction with 1.4.22rc5/1.6.1.1 |
| 20:20.08 | Qwell | Who dare make my IRC window flash? |
| 20:20.11 | dlynes | Qwell: _abc_'s asking about sccp/skinny |
| 20:20.11 | p3nguin | _abc_: I use SIP images on my 7900 series phones. |
| 20:20.14 | hardwire | Qwell: me |
| 20:20.21 | hardwire | dlynes: did you find a fixup? |
| 20:20.22 | _abc_ | how about skinny images? |
| 20:20.26 | Qwell | infobot: murder hardwire |
| 20:20.27 | infobot | ACTION shoots hardwire in his sleep |
| 20:20.33 | ManxPower-work | p3nguin: any sane person would be using SIP on Cisco phones. |
| 20:20.35 | hardwire | infobot: murder Qwell |
| 20:20.36 | infobot | ACTION shoots Qwell in his sleep |
| 20:20.36 | dlynes | hardwire: yeah...use sip, or upgrade the 1.4 side to 1.6 |
| 20:20.42 | hardwire | ok |
| 20:20.46 | dlynes | hardwire: or upgrade the 1.4 side to at least 1.4.26 |
| 20:20.57 | dlynes | hardwire: 1.4.25 and earlier had that issue |
| 20:20.59 | hardwire | well. your 1.4 side is my 1.2 side |
| 20:21.07 | p3nguin | I've actually considered going back to an SCCP image and fixing up chan_skinny on *. |
| 20:21.29 | dlynes | hardwire: my server that i was sending calls to was 1.4.22rc5 (elastix) |
| 20:21.37 | Qwell | p3nguin: what's to fix up? |
| 20:21.37 | _abc_ | so is skinny broken or not when used with cisco ? |
| 20:21.37 | dlynes | hardwire: my laptop was running 1.6.1.1 |
| 20:21.37 | hardwire | dlynes: is there any obvious change that made it not work? |
| 20:21.49 | dlynes | hardwire: there was a change that made it work...not that made it not work |
| 20:21.50 | p3nguin | qwell: I was under the impression that I would need to configure the conf for skinny before I could use it. |
| 20:22.21 | dlynes | hardwire: there's something different in the iax2 protocol in the way it authenticates between 1.4.25 and lower and 1.6.1.x |
| 20:22.37 | dlynes | hardwire: so 1.4.26 came out to address that issue (and probably other issues as well) |
| 20:23.05 | hardwire | yeh.. looks like it's rejecting long before it gets auth from the older box |
| 20:23.06 | p3nguin | _abc_: Is there any reason you don't want to convert the phones to SIP? |
| 20:23.18 | hardwire | I see it immediately rejecting the call.. then the older box sends the authentication |
| 20:23.19 | _abc_ | yes, i don't want to deal with that part yet |
| 20:23.37 | dlynes | hardwire: yeah, or something like that...i just remember it wasn't getting far enough to even show up on the console of the 1.4 box |
| 20:23.44 | dlynes | hardwire: unless i did a pcap |
| 20:23.57 | p3nguin | _abc_: It's the same amount of work to throw in the files for sccp as it is for sip. |
| 20:24.25 | _abc_ | p3nguin: you mean i need to futz with xml configs and sftp server? |
| 20:24.35 | _abc_ | eww i thought i could skip that step |
| 20:25.28 | dlynes | Anyone know where asterisk grabs the default periodic announcement file from for queues? |
| 20:25.34 | dlynes | Or is there a default file? |
| 20:25.59 | _abc_ | p3nguin: have you tried to run them on skinny before upgrading to sip images? |
| 20:26.09 | p3nguin | dlynes: queue-periodic-announce |
| 20:26.26 | p3nguin | _abc_: no, I just changed them to SIP as soon as I was ready to hook them up. |
| 20:27.12 | dlynes | p3nguin: thanks....i wonder why it can't find it in g729 format, then |
| 20:27.41 | p3nguin | Maybe you don't have it in that format? |
| 20:28.42 | dlynes | p3nguin: nvm....i followed someone else's instructions that said to set periodic-announce=ringing |
| 20:29.00 | dlynes | it's just my first time trying queues |
| 20:29.07 | p3nguin | ah |
| 20:29.59 | p3nguin | I didn't change any of the sound files in queues.conf, and I'm fairly satisfied with the results. |
| 20:30.56 | dlynes | p3nguin: How do you get it to play an announcement that they're getting put into the queue? Or do you have to do that as part of your dialplan? |
| 20:31.19 | p3nguin | I put it in the dialplan right before the Queue() command. |
| 20:31.21 | ecrist | does anyone know if I need to purchase a special license to change softkeys on polycom phones? |
| 20:31.38 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 20:32.01 | p3nguin | dlynes: Such as Playback(local/please-hold-for-agent), then Queue(myqueue). |
| 20:32.17 | dlynes | p3nguin: ok, so the queue doesn't have the feature, then |
| 20:33.17 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 20:33.18 | p3nguin | I can't say that it doesn't exist, but I don't do it that way. |
| 20:33.35 | *** join/#asterisk yoruk_ (n=yoruk@host164-182-dynamic.52-82-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 20:33.46 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) [NETSPLIT VICTIM] |
| 20:34.07 | *** join/#asterisk chai_sangeen (n=chai_san@84.255.164.36) |
| 20:36.29 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:38.20 | p3nguin | dlynes: With the default behavior of the queue, it starts playing MoH as soon as the queue is reached. It made sense to me to do a Playback() immediately before that to say to hold for a rep. Prior to that, callers are given a chance to dial internal phones if they know the extens. |
| 20:39.22 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
| 20:40.13 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 20:40.42 | dlynes | p3nguin: hrm....and then an added bonus |
| 20:41.11 | dlynes | p3nguin: in the sample file, it says you can use 'queue-lessthan=....', but asterisk doesn't recognize that keyword |
| 20:42.00 | p3nguin | You set queue-lessthan = queue-less-than in the queue and it chokes? |
| 20:42.21 | dlynes | p3nguin: yeah...throws an error when i module reload app_queue.so |
| 20:42.28 | dlynes | p3nguin: complains that's not a valid keyword |
| 20:42.41 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 20:42.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:42.41 | p3nguin | Did you check bug reports on that? |
| 20:42.53 | dlynes | p3nguin: checking issues.asterisk.org on it already....already beat you to it |
| 20:43.33 | p3nguin | I announce holdtime, but not lessthan. |
| 20:44.01 | p3nguin | I mean, I have announce-holdtime defined in my queue. |
| 20:44.04 | dlynes | p3nguin: i figured i'd try all the above |
| 20:44.17 | dlynes | and let asterisk figure it out what I wanted |
| 20:44.35 | *** join/#asterisk danj1980 (n=dan@91.108.0.113) |
| 20:44.41 | p3nguin | Now that I think about it, I don't know that I have ever heard it announce the time, so maybe that setting isn't doing any good for me. |
| 20:45.00 | dlynes | hehe |
| 20:45.00 | danj1980 | Hi everyone. |
| 20:45.22 | danj1980 | Just wondering, does anyone know what the licenses are needed for with Cisco 7940G phones? |
| 20:45.34 | danj1980 | Are they required if the phones are going to be connected to asterisk? |
| 20:45.38 | p3nguin | danj1980: No |
| 20:46.16 | danj1980 | Are there any features missing when you connect the phones to asterisk? ie. Are there any features that you only get when you connect the phones to Call Manager? |
| 20:46.17 | p3nguin | danj1980: You should have a SmartNet contract on your equipment, but if you're using SIP, I don't know what license you would be required to have. |
| 20:46.41 | p3nguin | Call Manager does do things that Asterisk doesn't, so yeah, some things are missing. |
| 20:46.47 | danj1980 | whats a smartnet contract? |
| 20:47.10 | p3nguin | http://www.cisco.com/en/US/products/svcs/ps3034/ps2827/ps2978/serv_group_home.html |
| 20:47.16 | danj1980 | got it |
| 20:47.18 | danj1980 | thanks |
| 20:47.28 | dlynes | p3nguin: aha! |
| 20:47.42 | dlynes | p3nguin: they changed the name of it, and didn't mention it in the queues.conf file |
| 20:47.59 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
| 20:48.19 | dlynes | p3nguin: it's now queue-reporthold |
| 20:48.27 | p3nguin | danj1980: If you don't ever need any support and you already have the SIP images, I don't see why you really _need_ to have a smartnet agreement. |
| 20:48.33 | dlynes | erm...nvm...that's a different one |
| 20:48.52 | dlynes | it seems they've removed the option |
| 20:49.00 | p3nguin | I... was just getting ready to mention that I saw both of those settings. |
| 20:49.15 | dlynes | at least in 1.6.1.8 it's been removed |
| 20:49.24 | dlynes | I'm looking at app_queue.c atm |
| 20:49.46 | p3nguin | I should enter the queue and hope it doesn't get answered, just to see what happens. |
| 20:50.19 | p3nguin | Actually, I'll create a new queue with no members. :) |
| 20:50.43 | *** join/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 20:50.57 | dlynes | p3nguin: it's been removed out of the sample file in 1.6.1.11, though |
| 20:51.55 | ManxPower-work | many telcos will drop the call if it's not answered in 120 seconds |
| 20:52.33 | dlynes | ManxPower-work: you mean many voip telcos, right? |
| 20:53.04 | danj1980 | p3nguin: thanks for your help |
| 20:53.40 | ManxPower-work | dlynes: no I mean all telcos |
| 20:53.45 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:53.47 | _abc_ | ok, thanks for the help all, i am going to go now |
| 20:53.58 | *** part/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 20:54.03 | p3nguin | I use VoIP.ms for my toll-free DID, and they have a setting for how long to let the phone ring. It goes all the way up to 300s. |
| 20:54.46 | p3nguin | Dial Time Out - The maximum amount of time a call to your DID can stay in "ringing state" before we cancel the call (No Answer). |
| 20:54.47 | ManxPower-work | p3nguin: Have you tested it at 300 seconds? |
| 20:55.06 | ManxPower-work | 'cause setting it in Asterisk won't change the way the PSTN telco carrier waits |
| 20:55.10 | p3nguin | No, because I answer in less than 1s. |
| 20:57.06 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 20:58.06 | danj1980 | p3nguin: Is it possible to download the SIP images? Anywhere? :-S |
| 20:58.36 | ManxPower-work | danj1980: I'm sure you can get the them the same place you get your pirated copies of Windows and Office and other software |
| 20:58.48 | p3nguin | ;) |
| 20:58.57 | ManxPower-work | danj1980: I assume you mean "SIP Images for Cisco" |
| 20:59.21 | p3nguin | Possible, yes. Cisco won't be fond of your doing it, though. |
| 21:00.29 | danj1980 | Are there any other phones that you can recommend that would be better, without any of these licensing issues? |
| 21:00.48 | [TK]D-Fender | Polycom > All |
| 21:00.56 | ManxPower-work | ~phones |
| 21:00.57 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 21:00.58 | danj1980 | We tried the Linksys SPA921 but its got firmware issues and they arent releasing any firmware updates. |
| 21:02.57 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 21:03.45 | danj1980 | Is it illegal to use an unlicenses cisco with asterisk? |
| 21:03.49 | danj1980 | unlicensed* |
| 21:04.40 | p3nguin | Technically, no. But they want you to have a contract in order to obtain the SIP firmware. |
| 21:04.48 | Corydon76-dig | It's a copyright violation with Cisco to use that phone with anything, not just Asterisk |
| 21:05.00 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:06.01 | Nugget | a "copyright violation" huh? |
| 21:06.07 | Corydon76-dig | The probability that Cisco will come after you, though, is low |
| 21:06.29 | Corydon76-dig | Nugget: yep, Cisco considers themselves a software company |
| 21:07.13 | Nugget | I'm not interesting in receiving legal advice from someone who doesn't even know the difference between copyright and licensing. |
| 21:07.38 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 21:07.44 | Corydon76-dig | I'm not giving you legal advice. Get that from your lawyer |
| 21:07.52 | dlynes | grrr....firefox and chatzilla are really starting to piss me off |
| 21:07.58 | danj1980 | Is there anyone who knows how to customise firmware? Maybe I can correct the faults in the SPA921 |
| 21:08.05 | Nugget | I mean, jesus fuck, you have a penguin tattoo. You're clearly a biased freetard. :) |
| 21:08.08 | angryuser_laptop | dlynes: use xchat |
| 21:08.20 | dlynes | angryuser_laptop: reason? |
| 21:08.38 | angryuser_laptop | dlynes: not being pissed off ? |
| 21:08.45 | ManxPower-work | I use pidgin for IRC. Only feature that I miss is not being able to /ignore someone |
| 21:09.04 | dlynes | angryuser_laptop: yeah..but i like chatzilla...it's just buggy lately...either that or it's firefox |
| 21:09.13 | dlynes | or maybe both |
| 21:09.25 | Corydon76-dig | Nugget: what you may not realize is that when I went in originally for the tattoo, the original plan was to get a Beastie tattoo |
| 21:09.30 | dlynes | it's just kinda crappy lately because chatzilla's a firefox addon now, instead of a separate program |
| 21:09.56 | dlynes | I actually like BitchX...just a bit of a pain to use it in a gui |
| 21:10.12 | Corydon76-dig | Nugget: I went for the simpler one first. Now I'm not so sure I want another. |
| 21:10.18 | p3nguin | BitchX is a pain to use ... in a gui? |
| 21:10.24 | angryuser_laptop | well xchas has a nice gui and it is free for lin/mac |
| 21:10.27 | p3nguin | Why would you need a gui for a command line IRC client? |
| 21:10.34 | angryuser_laptop | xchat* |
| 21:10.35 | dlynes | p3nguin: yeah..because you need to load up a terminal and then run it |
| 21:10.49 | p3nguin | Oh, well, I can see how xterm or aterm could get in the way of that. ;) |
| 21:10.55 | dlynes | p3nguin: because it's a great little irc client? |
| 21:11.05 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:11.12 | dlynes | I'm an ircii user from way back ;) |
| 21:12.17 | Corydon76-dig | Sirc is better |
| 21:12.57 | Nugget | GUIs are great for irc. They're just not as great as being able to run irc inside screen |
| 21:12.58 | p3nguin | I would prefer irssi over bx any day. |
| 21:13.08 | Nugget | and bitchx blows goats. |
| 21:13.14 | *** join/#asterisk CurtisKGwapo (n=CurtisKG@CPE00223f08d979-CM00223a6e9361.cpe.net.cable.rogers.com) |
| 21:13.29 | Nugget | all software sucks, but bitchx sucks on purpose. I've never encountered software that tries so hard to embarass its user as bitchx. |
| 21:13.40 | CurtisKGwapo | hey guys. Are there any unlimited sip trunk providers in Canada or the US? |
| 21:14.14 | ManxPower-work | ~siptrunk |
| 21:14.15 | infobot | i guess siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 21:14.42 | p3nguin | dlynes: I set up that test queue, and it says "Your call is now first in line..." immediately upon entering the queue. |
| 21:20.43 | DocAwesome | ircii ftw! |
| 21:21.42 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 21:22.48 | dlynes | Nugget: so i guess you hate epic then, too? |
| 21:23.16 | dlynes | p3nguin: test queue? |
| 21:23.51 | dlynes | p3nguin: i'm not getting the stupid thing to say squat, other than after 60 seconds, it pipes in and says i'm on hold and someone will be with me momentarily |
| 21:24.07 | dlynes | p3nguin: and then it continues to ring off the wall with music on hold |
| 21:24.28 | dlynes | good thing the caller can't see the phones continually ringing :) |
| 21:24.30 | p3nguin | dlynes: Oh, I thought I said that I was going to make a new queue with no members to test what happens when entering the queue. |
| 21:25.02 | dlynes | p3nguin: don't think so...sounds like a plan though, just so i can see what it does when everyone's busy |
| 21:25.28 | p3nguin | I defined those sounds, and it doesn't ever tell me hold time. It just says I am first in line and keeps me on hold. |
| 21:25.56 | dlynes | p3nguin: i don't think it's going to tell you hold time, unless it's had some calls, so that it's got a better idea how long it takes |
| 21:26.21 | p3nguin | I don't really mind, since my call volume is low anyway, and I only have a few people that will take calls at all. |
| 21:26.44 | dlynes | 'Your call is first in line. It shall be answered in approximately 5 days, 8 hours, 55 minutes, and 10 seconds. |
| 21:27.22 | DocAwesome | p3nguin: if you have "joinempty=yes" calls will enter without members, and "leavewhenempty=no" will let callers sit waiting for members if none are logged in (or available) |
| 21:27.31 | p3nguin | yeah |
| 21:27.54 | p3nguin | I had to specify "joinempty=yes" to be able to call into this test queue which as no members. |
| 21:27.55 | DocAwesome | dlynes: I think by default it rounds up to the nearest minute :) |
| 21:27.57 | [T]ank | ive been trying to get a polycom soundstation ip 6000 set up with my asterisk server... I have created an ftp server and all of the files, but am getting a configuration error. anyone willing to work with me on setting this up correctly? |
| 21:28.24 | dlynes | DocAwesome: and leavewhenempty=yes, it just hangs up on you with no warning, when there's no members in the queue? |
| 21:28.40 | DocAwesome | dlynes: I think it continues to the next priority |
| 21:28.46 | DocAwesome | i.e. send to voicemail |
| 21:28.52 | dlynes | DocAwesome: ok, thanks |
| 21:29.03 | p3nguin | Even before the queue() timeout is reached? |
| 21:29.04 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 21:29.07 | DocAwesome | if not, then I'm pretty sure there is an option to do that |
| 21:29.12 | DocAwesome | p3nguin: yes |
| 21:29.45 | DocAwesome | Queue() in asterisk is actually pretty slick. There have been very very few things I can't do that have been requested by any of my clients |
| 21:29.55 | DocAwesome | I can't even think of one, that's how few there are! :) |
| 21:31.41 | [T]ank | here are my configs both for the polycom which are on the ftp server and the sip.conf http://pastebin.ca/1708782 |
| 21:32.13 | [T]ank | could anyone please help me figure out why i am getting a "config error" |
| 21:32.36 | [T]ank | Config File Error 0x20 |
| 21:33.26 | ChannelZ | pastebin appears to be dead |
| 21:33.44 | Godfather_ | for me doesnt work pastebin.ca |
| 21:34.02 | [T]ank | oh, nice... it must have just died. |
| 21:34.12 | [T]ank | let me see if i can recover and use a different site |
| 21:34.27 | ChannelZ | your config is so screwed it trashed pastebin |
| 21:34.39 | [T]ank | HAHA, probably :-D |
| 21:34.42 | ChannelZ | there's pastebin.com |
| 21:34.50 | *** join/#asterisk leoburd_ (n=leoburd@wireless-25-40.media.mit.edu) |
| 21:34.52 | Godfather_ | jaja |
| 21:34.55 | Godfather_ | now works! |
| 21:34.56 | ChannelZ | ah ca just finally came up |
| 21:35.01 | [T]ank | ok |
| 21:35.13 | [T]ank | good, cuz the past of this had line numbers and crap in it. was gonna be a mess |
| 21:35.28 | ChannelZ | [T]ank: what version of * |
| 21:36.05 | [T]ank | 1.6.1.1 |
| 21:36.14 | ManxPower-work | config error on polycoms almost always means mismatched " in the polycom config file. |
| 21:36.36 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:37.07 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:37.12 | leoburd_ | NEWBIE QUESTION: hello there, I'm getting a 'file does not exist' error every time I try to execute my agi script... but the file seems to be there... what shall I do? |
| 21:38.00 | p3nguin | leoburd_: Check the permissions on the script file. |
| 21:38.02 | [T]ank | leoburd_ have you checked the permissions and such on the file you are trying to use? Does asterisk have rights to execute it? |
| 21:38.08 | ChannelZ | does the user asterisk is running as have permission to read/execute the file? |
| 21:38.33 | [T]ank | wow... seems that would be the first thing to check ;-) |
| 21:39.11 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
| 21:39.38 | p3nguin | The problem I ran into the other day was that /usr/local/bin/php was the interpreter specified in the shebang inside the script, but that wasn't the path to php. It gave the same result as if the script didn't exist. |
| 21:39.52 | ChannelZ | she bangs! she bangs! |
| 21:40.26 | leoburd_ | let me check... |
| 21:42.28 | ChannelZ | [T]ank: so you have a new problem on the phone side now, not * ? |
| 21:42.45 | [T]ank | You mean from what I was trying to figure out last night? |
| 21:43.17 | [T]ank | because i was using the web interface, I was assuming that it was not setting something that asterisk was wanting.... so i set up an ftp server and am trying it this way where i have more access to the configs. |
| 21:43.19 | ChannelZ | yeah |
| 21:43.31 | [T]ank | Now I cant even get it to boot up. I have something in the configs set up wrong. |
| 21:43.41 | [T]ank | ive been at this for 3 days now :-D |
| 21:43.44 | [T]ank | Im persistant. |
| 21:43.51 | ChannelZ | as Manx said probably a syntax error somewhere but that thing is such a mess of text my eyes are crossing just looking at it |
| 21:44.10 | ChannelZ | But don't try to provision it right now. Get it to work yourself. Factory reset the thing and use the web interface |
| 21:44.13 | [T]ank | here is what I am using for tutorial. http://www.sureteq.com/asterisk/polycom.htm#5.%C2%A0_Polycom_configuration_files_ |
| 21:46.04 | *** join/#asterisk yoshx (n=yoshx@78.114.253.27) |
| 21:46.46 | *** join/#asterisk vally (i=vally@ip-92-50-113-160.unitymediagroup.de) |
| 21:47.29 | Katty | hmm. |
| 21:47.34 | Katty | i'm thinking about going home early |
| 21:50.23 | [T]ank | ChannelZ: yeah, thats where i started.... I will do that. Then put together some screenshots. back in a few |
| 21:52.11 | Kobaz | early? that's blasphemy |
| 21:52.37 | ChannelZ | [T]ank: I don't run * 1.6 so I only have two theories based on what was going on last night: 1. Your Polycom is sending an Authorization line when it shouldn't because none of that stuff is filled (so it's trying to auth with a blank username). Why, I don't know, maybe it's a bug and you need new firmware or something. 2. The 'username' directive in your sip.conf is making * 1.6 act as if it requires authoization - 1.4 doesn't do this but maybe 1.6 d |
| 21:52.38 | ChannelZ | oes. |
| 21:53.27 | [TK]D-Fender | ~polycomprovisioning |
| 21:53.28 | infobot | People who configure Polycom phones via the web interface or via the phone itself should be dragged out and shot. Survivors should be shot AGAIN. |
| 21:53.50 | Kobaz | haha |
| 21:54.10 | [T]ank | [TK]D-Fender nice to be loved. |
| 21:57.58 | Katty | has chili |
| 21:59.12 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 21:59.12 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 21:59.29 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:59.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:59.31 | *** join/#asterisk ticoit (n=ticoit@201.191.190.123) |
| 22:00.05 | hardwire | meh |
| 22:00.12 | Katty | why |
| 22:00.21 | hardwire | ITSP can eat me. |
| 22:00.25 | Katty | k |
| 22:00.39 | Katty | <ITSP> omnomnomnom |
| 22:00.47 | hardwire | they change stuff on their side then wonder why I'm not doing it right. |
| 22:00.48 | hardwire | it's annoying |
| 22:01.06 | Katty | that's happened to me. |
| 22:01.19 | Katty | one time, charter changed a public ip address on a client of ours |
| 22:01.23 | Katty | didn't tell anyone. |
| 22:04.11 | DocAwesome | Katty: CHILI?! |
| 22:04.13 | DocAwesome | <-- wants |
| 22:04.49 | dlynes | Katty: don't stick your bum anywhere near me with that chili |
| 22:04.50 | x86 | mmmm chili |
| 22:04.56 | hardwire | ... |
| 22:04.59 | x86 | mmmmmm bum |
| 22:05.04 | x86 | lulz |
| 22:05.07 | *** join/#asterisk flux_control (n=flux@sourcemage/wizard/flux) |
| 22:05.45 | dlynes | Katty: we get that sorta crap happening on a regular basis with our stupid isp |
| 22:06.21 | flux_control | I have asterisk 1.4.21.2 installed, but I'm having a problem with logging. I can't get asterisk to log either to regular files or to syslog, defined in logger.conf. Is there a module I should be loading to enable logging? |
| 22:07.19 | dlynes | flux_control: I seem to remember having a problem with a 1.4 version around that vintage with that same problem, too...figured it was just me, or the way someone set it up....maybe i was wrong.... |
| 22:07.30 | flux_control | I see. |
| 22:07.52 | [T]ank | ChannelZ resetting to factory default does not seem to completely work.... its still complaining of a config error, even though all of the settings on the phone show that its not pointed to an ftp server at all |
| 22:07.55 | dlynes | flux_control: I could've sworn it was 1.4.22rc5 or something though...but don't remember |
| 22:08.06 | flux_control | I'm planning on upgrading to the latest 1.6.1.11 anyway, but I figured I'd check to see if I was just missing something first. |
| 22:08.11 | dlynes | flux_control: i've since upgraded all those boxes to 1.4.26.2 so it's not an issue |
| 22:08.33 | flux_control | Right, that's what I'm hoping for myself as well. :) |
| 22:08.53 | dlynes | flux_control: do you have multiple boxes that need to talk to each other via iax2? |
| 22:09.00 | flux_control | Nope |
| 22:09.21 | flux_control | Just one box running asterisk, and clients which connect. |
| 22:09.23 | dlynes | flux_control: ok...just asking, because 1.4.25 and earlier couldn't talk iax2 to 1.6.1 boxes |
| 22:09.38 | diatonic | [T]ank: Can you do the 'hold down * 4 6 & 8' for 5 seconds to wipe that thing? |
| 22:09.41 | flux_control | Ah, thanks for the heads up :) |
| 22:09.41 | ChannelZ | [T]ank: well I dunno what to tell you there if the thing has an error with a default config |
| 22:09.56 | dlynes | flux_control: but if they're all doing sip, there isn't an issue |
| 22:10.06 | flux_control | I'm also having issue with using an ipkall DID I set up. |
| 22:10.12 | diatonic | [T]ank: I think that also formats the filesystem |
| 22:10.24 | flux_control | I've searched about it, but it seems inconclusive as to whether ipkall is still working fine or not. |
| 22:10.28 | dlynes | flux_control: yeah...i think you were on last night under a different nick asking about that, right? |
| 22:10.40 | flux_control | dlynes: No, that wasn't me. |
| 22:10.42 | danj1980 | Hi, does anyone have a wholesale distributor for the Aastra 480I IP Phone, in the UK? |
| 22:10.45 | dlynes | flux_control: ah |
| 22:10.58 | [T]ank | diatonic: docs i found said just *,8,6. I will try yours |
| 22:12.30 | flux_control | I can receive the call, and then I can dial my sip phone (soft phone currently), but no audio goes between them. Also, the SIP phone terminates connection after about 20sec, but the PTNS continues to be connected until it reaches the ipkall timeout. |
| 22:13.04 | flux_control | Dialing echo tests works with the soft phone though, so I don't think it's a nat issue (at least not on my end), else I should have audio problems there too, right? |
| 22:13.57 | flux_control | Kind of hard to debug it without having logging though.. :( |
| 22:14.15 | flux_control | I'm wondering if that issue will magically go away after the upgrade as well.. |
| 22:14.44 | Katty | gives DocAwesome chili. |
| 22:14.54 | flux_control | Just wondering if anyone else has * currently working with ipkall DIDs or not. |
| 22:15.05 | Katty | i don't |
| 22:15.09 | diatonic | i don't |
| 22:15.15 | ChannelZ | I like peanut butter |
| 22:15.21 | Katty | i do too |
| 22:15.27 | diatonic | i don't |
| 22:15.30 | [T]ank | are you behind nat? |
| 22:15.34 | flux_control | Alrighty |
| 22:15.38 | [T]ank | flux_control are you behind nat? |
| 22:15.40 | Katty | well you're just weird. |
| 22:15.47 | diatonic | okay, i do |
| 22:15.52 | Katty | k |
| 22:16.02 | ChannelZ | but I am suspicious of a company who can't spell "call" right |
| 22:16.07 | DocAwesome | I like organic PB and not things like Jiffy |
| 22:16.17 | Katty | i like smucker's natural creamy |
| 22:16.17 | flux_control | [T]ank: My * box is in the DMZ, but the clients are behind nat. However, I have canreinvite=no, so the calls get routed through the * box. |
| 22:16.18 | DocAwesome | the no-stir organic tastes a million times better |
| 22:16.29 | ChannelZ | JIF all the way baby |
| 22:16.32 | Katty | tho i'm not sure if it's organic or not |
| 22:16.44 | dlynes | danj1980: have you Netco? |
| 22:16.45 | flux_control | It could be my nat, but I thought that if nat was the cause the echo tests would fail too. |
| 22:16.55 | dlynes | s/Netco/tried Netco/ |
| 22:17.03 | Katty | have you tried nacho |
| 22:17.09 | ChannelZ | mmmm nachos |
| 22:17.11 | [T]ank | flux_control: no idea... what ports do you have forwarded? |
| 22:17.14 | flux_control | Nachos are tasty :) |
| 22:17.23 | diatonic | I only like the PB found inside Reese's peanut butter cups. To make a sandwich I have to hollow out about 12 Reece's PB cups |
| 22:17.24 | flux_control | [T]ank: Forwarded from where to where? |
| 22:17.29 | Katty | they totally are. |
| 22:17.55 | dlynes | Katty: smuckers anything isn't organic |
| 22:18.11 | [T]ank | flux_control: exactly. If you are behind nat, what ports are you forwarding from the internet to your asterisk server |
| 22:18.22 | dlynes | Katty: they were one of the ones that got caught in that tainted peanut scandal a couple of years ago |
| 22:18.25 | flux_control | [T]ank: No, the * server is in the DMZ. |
| 22:18.37 | Katty | dlynes: http://www.bigapplegrocer.net/ProdImages/20999.jpg <- orly |
| 22:18.46 | flux_control | [T]ank: My setup is CableModem -> router -> Asterisk Server + Clients. |
| 22:18.55 | dlynes | Katty: woah...that's new |
| 22:19.03 | dlynes | Katty: Maybe it's stateside, only? |
| 22:19.05 | [T]ank | flux_control: so no nat then |
| 22:19.09 | flux_control | The * server is in the DMZ, so it sees everything incoming. |
| 22:19.13 | Katty | dlynes: who knows |
| 22:19.16 | dlynes | Katty: I've never seen it in Canada |
| 22:19.33 | diatonic | I would think organic would be more likely to be contaminted with bad stuff |
| 22:19.42 | flux_control | The clients are not in the DMZ, so they only see associated incoming, but they can communicate directly with the asterisk server. |
| 22:19.45 | dlynes | diatonic: like ddt? |
| 22:19.48 | ChannelZ | [T]ank: DMZ is like still behind NAT but with every port forwarded |
| 22:19.51 | diatonic | no, like bacteria |
| 22:20.05 | diatonic | e coli & whatnot |
| 22:20.20 | dlynes | diatonic: bacteria gets into organic the same way it gets into non-organic |
| 22:20.28 | dlynes | diatonic: so i fail to see your point |
| 22:20.47 | flux_control | In the * server, I set nat=no for the clients (since wrt the * box they are not nat), and canreinvite=no so that communication with anything is forced through the * box (which is for all purposes not nat anywhere). |
| 22:21.02 | diatonic | non-organic can treat with anti-bacterial agents. Organic can not |
| 22:21.06 | flux_control | At least that's how I understand how the config options in * work. |
| 22:21.14 | dlynes | diatonic: huh? |
| 22:21.37 | diatonic | dlynes: nevermind - way off topic |
| 22:21.39 | dlynes | diatonic: you mean like sulfates? |
| 22:22.12 | dlynes | organic means it doesn't use pesticides...has nothing to do with bacteriacides |
| 22:22.49 | Katty | well |
| 22:22.53 | Katty | you can still use some pesticides |
| 22:22.59 | flux_control | So, outside of ipkall, does anyone have a working DID (USA) with asterisk? |
| 22:23.11 | flux_control | Preferably a free one... |
| 22:23.11 | Katty | it's just greatly restricted |
| 22:23.19 | flux_control | I'd like to "try before I buy" so to speak ;) |
| 22:23.37 | Katty | the best way to get organic food is to just grow it in your back yard |
| 22:24.05 | ChannelZ | The Snickers seeds I planted never came up |
| 22:25.16 | dlynes | flux_control: vitelity.net |
| 22:25.37 | dlynes | flux_control: that's what I've used to get some USA dids |
| 22:26.04 | flux_control | dlynes: I'm checking them out now, thanks. |
| 22:26.10 | dlynes | flux_control: mind you, it doesn't work the way they say it does |
| 22:26.16 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:26.17 | dlynes | flux_control: you've gotta use some common sense |
| 22:26.34 | diatonic | I'm using vitelity for DIDs. Aside from some DTMF issues, they've been pretty good |
| 22:26.57 | *** join/#asterisk jtodd (i=k16uqwxr@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 22:26.57 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:26.58 | dlynes | diatonic: yeah..i've only ever used them for inbound dids...never used them for outbound service |
| 22:27.13 | jdnwest | Anyone else have problems with Asterisk NOW installing broken? |
| 22:27.32 | dlynes | jdnwest: guis are broken in general...can you be specific? |
| 22:27.39 | diatonic | dlynes: Same here. All of our outbound gets carried on TDM hardware |
| 22:27.50 | flux_control | dlynes: Can you be more specific about "it doesn't work the way they say it does"? |
| 22:27.58 | jdnwest | retrieve_conf failed, config not applied |
| 22:28.09 | flux_control | I'm only looking for inbound DID, not outbound service. |
| 22:28.10 | jdnwest | basically the linking between the webconfig, and the actual files seemed to be broken. |
| 22:28.16 | dlynes | flux_control: i had to make some changes to the sip config |
| 22:28.22 | flux_control | Ah |
| 22:28.34 | dlynes | flux_control: i..e their example sip config didn't work |
| 22:28.36 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:28.40 | flux_control | Anything that wouldn't be straightforward? |
| 22:28.54 | dlynes | flux_control: yeah..pretty straightforward...it wasn't anything hokey |
| 22:29.07 | flux_control | AFAIK, if I have allowguests=yes I shouldn't even need anything in sip.conf, just a dialplan for the connection, right? |
| 22:29.25 | dlynes | flux_control: one sec |
| 22:29.37 | flux_control | This isn't on a production asterisk machine, and it doesn't have outbound calling anywhere, so for now I don't mind allowing guest sip connections. |
| 22:30.32 | jdnwest | Flux_Control: I apreciate the work the deveopers do, but i feel like asteriskNOW tarnishes Digium's/Asterisks credibility and they should just put a link to one of the projects that actually works. |
| 22:30.45 | dlynes | flux_control: you need a register, complete with a port number (username:secret@66.241.96.96:5060) (if I remember correctly, that wasn't the ip address they gave me, either) |
| 22:31.29 | dlynes | flux_control: and then you also need a [vitelity-inbound] sip context |
| 22:31.39 | diatonic | flux_control: You can do it based on IP authentication, and Vitelity provides changes you need to make to sip.conf and extensions.conf |
| 22:31.43 | flux_control | jdnwest: Wrong tab-completion? |
| 22:31.44 | dlynes | flux_control: with a host of inbound23.vitelity.net, defined as type friend |
| 22:32.22 | dlynes | flux_control: and insecure=port,invite (so, yes...this was the one weird thing) |
| 22:32.23 | flux_control | dlynes: Did you ever try it with allowguests=yes in sip.conf? |
| 22:32.30 | dlynes | flux_control: never |
| 22:32.32 | flux_control | Just curious |
| 22:32.49 | flux_control | Also, it seems they don't offer any free DIDs, correct? |
| 22:32.52 | jdnwest | flux_control: I just copy/pasted the sip.conf that's under the asterisk support tab. |
| 22:33.01 | drfreeze | Anyone know what a voicemail light wouldn't light up on a Polycom phone? |
| 22:33.03 | dlynes | flux_control: no...i think you want ipkall or something for that |
| 22:33.04 | jdnwest | No, but at a 1.50 a month, its about as close as you can get. |
| 22:33.05 | DocAwesome | Asterisk Release Candidates are now available: 1.4.28-rc1, 1.6.0.20-rc1, 1.6.1.12-rc1, and 1.6.2.0-rc8. Please see the release announcement at http://www.asterisk.org/node/49875. Thanks! |
| 22:33.19 | drfreeze | I'm using the same setup I have done recently, and no lights show up |
| 22:33.22 | flux_control | jdnwest: No, I meant did you mean that message for someone else? I don't know anything about asterisk now, or why you were telling me. :/ |
| 22:33.34 | flux_control | Yeah, 1.50 is pretty cheap |
| 22:33.46 | jdnwest | Flux:lol, general venting in this channel. no one took my flame bait |
| 22:33.49 | dlynes | flux_control: dlynes is pretty close to flux_control , i think...don't you? |
| 22:33.52 | flux_control | I'd just like to make sure my setup is definitely working properly before I start shelling out cash. |
| 22:33.58 | diatonic | DocAwesome: Access denied to that URL :( |
| 22:34.07 | DocAwesome | oh I know why! |
| 22:34.10 | Katty | i'mmmmmmmmmmmmmmm int he mooood for love |
| 22:34.16 | flux_control | dlynes: lol, well, f and d on a qwerty board are right next to each other ;) |
| 22:34.46 | dlynes | flux_control: oh yeah...he could've hit 'fl<tab>', instead of 'dl<tab>' |
| 22:34.58 | flux_control | I also tryed sipbroker. |
| 22:35.18 | dlynes | jdnwest: have you tried asking in #asterisknow? |
| 22:35.21 | jdnwest | flux_control: Only problem i've ever had with vitelity was some DTMF issues once going to my apartment's gate box. Their supports good, and my voicemail vendor recommends them now for trunking to his boxes (not asterisk). |
| 22:35.28 | Katty | simply because you're neaaaaaaar meeeee!!! |
| 22:35.31 | dlynes | jdnwest: it's just freepbx and asterisk on centos |
| 22:35.34 | flux_control | Audio works for that one, but the connection still dies on the local (called) side after 10-20 secs, while the remote (calling) side stays connected. |
| 22:35.42 | DocAwesome | diatonic: try now |
| 22:35.53 | flux_control | jdnwest: Did you try different dtmf specifications? Inband, etc.? |
| 22:35.55 | DocAwesome | forgot to his "Published" |
| 22:35.58 | dlynes | DocAwesome: still denied |
| 22:36.02 | DocAwesome | really... |
| 22:36.03 | DocAwesome | that's weird |
| 22:36.28 | dlynes | really....maybe the publish date is in the future? |
| 22:36.50 | diatonic | I had to set dtmf to info or inband with vitelity |
| 22:36.52 | jdnwest | Dlynes:lol, yah, that channel is dead, If you awnt freepbx on centos PBXinaFlash and Elastix both actually do that. I've installed something else though and my rage at digium is dieing down. |
| 22:37.05 | flux_control | DocAwesome: Directory permissions problem? Url permission problem in the http server? |
| 22:37.22 | dlynes | flux_control: could be a chmod or chown issue, too |
| 22:37.25 | DocAwesome | nothing should be wrong as this is exactly what I've done with all the other release announcements |
| 22:37.35 | diatonic | Drupal cache issue? |
| 22:37.42 | DocAwesome | it's highly unlikely it's a permissions problem unless the webmaster effed something up |
| 22:37.47 | flux_control | Drupal... *shiver* |
| 22:37.50 | flux_control | :) |
| 22:38.01 | diatonic | Drupal = awesomesauce |
| 22:38.02 | drfreeze | Any polycom voicemail experts out there? |
| 22:38.07 | dlynes | DocAwesome: something is wrong, though...maybe someone changed something on your drupal without telling you? |
| 22:38.10 | bcrisp | What type of manager event should i be looking for when a queued caller's position is updated? |
| 22:38.12 | flux_control | DocAwesome: When in doubt, blame the webmaster. ;p |
| 22:38.22 | DocAwesome | well there is something wrong with that story it seems |
| 22:38.26 | jdnwest | who then blames the CRM.... |
| 22:38.32 | DocAwesome | I think I will just delete that one and make a new one |
| 22:38.36 | jdnwest | CMS* |
| 22:38.56 | dlynes | dood....wordpress kicks drupal's ass ;) |
| 22:39.20 | diatonic | dlynes: wow |
| 22:39.38 | flux_control | Anyone know why asterisk would drop a sip call prematurely? |
| 22:39.48 | flux_control | Or perhaps it's my soft phone? (linphonec) |
| 22:39.52 | dlynes | diatonic: mostly because wordpress is dead simple....drupla's a bit complicated :0 |
| 22:39.59 | flux_control | Though I get the same issue with SJPhone on Mac. |
| 22:40.06 | dlynes | flux_control: could be a network issue |
| 22:40.26 | flux_control | dlynes: That's what I was thinking, but I couldn't think of what, unless it was on the remote end. |
| 22:41.04 | dlynes | flux_control: could be port on the switch, could be network cable, could be interrupts on the card, could be the network card driver, could be.... |
| 22:41.08 | flux_control | I did see something online about an older version of * violating the rfc regarding how it was sending the ack, but like I said for an older version which supposedly got fixed well before the version I have. |
| 22:41.42 | flux_control | dlynes: If it was my card/cable/etc., should I experience the same drop when I do outbound calls from the softphone, like to remote echo tests? |
| 22:41.54 | flux_control | For example, I can call echo@iptel.org and have no problem. |
| 22:42.06 | DocAwesome | wow, that was heavily annoying |
| 22:42.15 | DocAwesome | Asterisk Release Candidates are now available: 1.4.28-rc1, 1.6.0.20-rc1, 1.6.1.12-rc1, and 1.6.2.0-rc8. Please see the release announcement at http://www.asterisk.org/node/49876. Thanks! |
| 22:42.19 | diatonic | drfreeze:Is your Polycom with VM problems being provisioned from a server? |
| 22:42.21 | flux_control | If I get the logging working I can probably debug it all myself. |
| 22:42.26 | flux_control | Darn logging :/ |
| 22:43.01 | DocAwesome | flux_control / dlynes / diatonic: give it a shot now |
| 22:43.08 | DocAwesome | note the node number has increased by 1 |
| 22:43.25 | diatonic | DocAwesome: New node works |
| 22:43.34 | DocAwesome | no idea what was wrong with the old one |
| 22:44.01 | flux_control | Yup, here too. |
| 22:44.14 | flux_control | DocAwesome: The good old fence-post problem? :) |
| 22:45.07 | flux_control | Hmm.. does anyone know of any other test numbers? |
| 22:45.20 | flux_control | I want to make sure I'm not having a general call problem... |
| 22:45.48 | flux_control | If there's an echo test or something that will allow 40 seconds or show that should be sufficient. |
| 22:45.49 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 22:46.15 | ChannelZ | http://www.voip-info.org/wiki/view/Phone+Numbers |
| 22:46.35 | flux_control | I know those. |
| 22:46.40 | drfreeze | diatonic: yes |
| 22:46.55 | flux_control | Not so many echo tests, and the ones listed don't seem very long. |
| 22:47.25 | drfreeze | I just finished an office with the same setup (except this time used the latest polycom config files. Did not do a straight copy) |
| 22:48.26 | diatonic | drfreeze: If you pastebin the macaddr.cfg and macaddrreg.cfg I'll see if anything rumps out at me as to why MWI isn't working |
| 22:48.41 | diatonic | err... jumps out at me |
| 22:49.10 | ChannelZ | the ekiga echo is letting me go on and on |
| 22:50.01 | flux_control | I'll try that one, usually I get denied with ekiga ones |
| 22:50.32 | Katty | DENIED |
| 22:50.32 | ChannelZ | sip:*010600@ekiga.net |
| 22:50.39 | Katty | goes home |
| 22:50.49 | flux_control | ChannelZ: I'm trying that one. |
| 22:51.00 | flux_control | That echo isn't working for me... |
| 22:51.05 | flux_control | I get no audio. |
| 22:51.22 | flux_control | Yet I get audio from other echo tests (like echo@iptel.org), and the TELLME calls work. |
| 22:51.59 | ChannelZ | You get nothing from them or just nothing back on the echo? |
| 22:52.54 | flux_control | Either |
| 22:53.02 | flux_control | I hear nothing at all. |
| 22:53.17 | *** join/#asterisk brozow (n=brozow@nc-63-162-204-51.sta.embarqhsd.net) |
| 22:53.20 | ChannelZ | hmm well I dunno whats going on through your screwy system |
| 22:53.28 | flux_control | lol |
| 22:53.34 | ChannelZ | This is a softphone on the same LAN as your * box calling that number right? |
| 22:53.43 | flux_control | Yup |
| 22:53.51 | flux_control | Routing through the * box. |
| 22:54.11 | ChannelZ | what happens if you make an extension in your * that does Dial(SIP/ekiga.net/*010600) and call that extension from the softphone? |
| 22:54.44 | flux_control | I'll try that. |
| 22:54.51 | ChannelZ | what softphone is this btw |
| 22:55.03 | flux_control | linphonec |
| 22:55.13 | flux_control | Interesting... |
| 22:55.28 | ChannelZ | and you say you're going 'through' the * box, but if you're typing in a direct SIP address, is it *really* going through the * box at all? |
| 22:55.29 | flux_control | I just did another call to echo@iptel.org and saw some errors in my softphone client. |
| 22:56.17 | flux_control | Things along the lines of "ortp-error-Payload telephone-event type already entered, should not happen !" |
| 22:56.45 | flux_control | ChannelZ: It's not allowed to do reinvite, and the * box is used as the registrar, proxy, gateway, etc. |
| 22:58.09 | ChannelZ | and the call is actually going through * on the console |
| 23:00.27 | flux_control | What do you mean going through * on the console? |
| 23:00.43 | flux_control | You mean using the * console originate to place the call instead of the softphone? |
| 23:00.48 | ChannelZ | when you make this call from the softphone you see the activity of * being involved? |
| 23:00.59 | flux_control | Where? |
| 23:01.04 | flux_control | Please be more specific |
| 23:01.11 | drfreeze | diatonic: http://pastie.textmate.org/private/ceeqatwjydwgfzpjmnjqg |
| 23:01.11 | ChannelZ | Like what does your dialplan look like that you're making SIP calls from the softphone |
| 23:01.15 | ChannelZ | yes on the console |
| 23:01.51 | flux_control | If I turn on sip debug in the asterisk console, I can see all the packets, including audio packets. |
| 23:02.18 | flux_control | When my * box receives a call, it dials the sip extension that the softphone registers to. |
| 23:03.44 | ChannelZ | but your outgoing calls |
| 23:04.22 | diatonic | drfreeze: What model polycom? |
| 23:04.33 | flux_control | I get no audio using an extension in the dialplan for Dial(SIP/ekiga.net/*010600) either. |
| 23:04.37 | ChannelZ | to these SIP test echos and things... I do not think * behaves in the way you think it behaves in this case, maybe I'm wrong |
| 23:05.26 | flux_control | ChannelZ: When I place an outbound call to echo@iptel.org, and I have sip debug on, I see all the packets in the console (including audio). |
| 23:05.26 | *** part/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 23:05.44 | flux_control | That's for when I place a call on my softphone directly to iptel. |
| 23:05.51 | ChannelZ | well audio doesn't go through SIP first of all so I dunno what you're going on about there |
| 23:06.07 | flux_control | There's no such thing as a "direct SIP call", unless the pathway between the two endpoints allows for it. |
| 23:06.25 | flux_control | ChannelZ: Signalling for the audio, not the audio itself. |
| 23:06.47 | flux_control | And I didn't say that the audio was going through SIP. |
| 23:07.07 | ChannelZ | ... " I have sip debug on, I see all the packets in the console (including audio)." |
| 23:07.23 | ChannelZ | but whatever. I have no idea what craziness you have going on, I will cease trying to help |
| 23:07.26 | flux_control | Yes, the packet headers which refer to the audio. |
| 23:07.42 | ChannelZ | I need to go swap HDs in a computer. |
| 23:07.48 | flux_control | Anyway, iptel.org works fine, they just don't allow it to go on forever. |
| 23:07.58 | flux_control | So I can't do lengthy tests with them. |
| 23:08.19 | ChannelZ | they seem not to be your problem |
| 23:10.48 | flux_control | ChannelZ: "they" = ??? |
| 23:11.03 | flux_control | iptel? |
| 23:13.13 | flux_control | Hmm..when looking at a packet, which one initiates the BYE? --v |
| 23:13.14 | flux_control | Really destroying SIP dialog '680793495dd83eca04fdc71c5fa741ea@192.169.1.3' Method: INVITE |
| 23:13.17 | flux_control | Really destroying SIP dialog '3e92812628bb2bac03a957966ab8202f@192.169.1.3' Method: BYE |
| 23:13.22 | flux_control | Sorry, wrong paste. |
| 23:13.24 | flux_control | <--- SIP read from 213.192.59.75:5060 ---> |
| 23:13.25 | flux_control | BYE sip:asterisk@72.231.218.64:1036 SIP/2.0 |
| 23:13.28 | flux_control | There we go. |
| 23:14.00 | flux_control | Is the BYE initiated from the 72.*, or from the 213.*? From the 213, right? |
| 23:14.22 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 23:15.08 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733516.dsl.bell.ca) |
| 23:15.53 | Qwell | flux_control: the answer is in the first line |
| 23:15.55 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 23:16.18 | flux_control | Qwell: Answer to my question, or answer to a SIP request? |
| 23:16.23 | dwery | hello, anyone is using the latest dahdi with kernel 2.6.32? |
| 23:16.25 | flux_control | Just want to be sure. |
| 23:16.33 | Qwell | the answer to where the packet came from |
| 23:16.39 | flux_control | OK |
| 23:16.52 | flux_control | So it's a BYE packet sent to 72.* then |
| 23:16.57 | flux_control | That's good then :) |
| 23:18.52 | flux_control | Oh, I'm also a little confused wrt SIP presence support in asterisk. |
| 23:19.07 | flux_control | It seems like it should be supported in 1.6, and there may have been a backport to 1.4? |
| 23:19.13 | *** join/#asterisk gooph (n=gooph@pool-71-96-244-205.dfw.dsl-w.verizon.net) |
| 23:19.29 | flux_control | Am I mistaken about this? If there is support in 1.6, is it "complete"? |
| 23:20.49 | flux_control | Is there another way to find out a devices state before actually calling the device? |
| 23:26.48 | flux_control | By the way, I meant the PUBLISH method of announcing presence. Sorry. |
| 23:27.07 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 23:28.01 | *** join/#asterisk BuSyAnToS (n=31749@93-44-17-172.ip95.fastwebnet.it) |
| 23:29.05 | *** join/#asterisk dan__t (i=vpn@vpn.withparity.net) |
| 23:29.16 | dan__t | I love AEL. |
| 23:30.26 | dan__t | Anyone else use it a lot? |
| 23:30.48 | Kobaz | yes |
| 23:30.51 | Kobaz | it's sexy |
| 23:30.58 | Kobaz | although i've been moving much of my ael to agi |
| 23:31.03 | flux_control | I use extensions.ael in lieu of extensions.conf entirely. |
| 23:31.09 | Kobaz | yeap |
| 23:31.15 | *** part/#asterisk BuSyAnToS (n=31749@93-44-17-172.ip95.fastwebnet.it) |
| 23:31.17 | Kobaz | why would you want to go through the pain of writing in BASIC |
| 23:31.41 | Kobaz | 10,print foo |
| 23:31.43 | Kobaz | 20, goto 10 |
| 23:32.44 | bcrisp | sounds neat |
| 23:33.01 | bcrisp | looking at ael now.. |
| 23:33.51 | bcrisp | Kobaz.. got a question about qeueus. Is it possible for a caller to leave a voicemail rather than waiting for the agent, and have that message stay in the queue in their stead? |
| 23:34.02 | Kobaz | context foo { s => { Answer(); while (1) { Playback(lunch); } } } |
| 23:34.37 | dan__t | I'm still not that good of a programmer to use AGI yet, but I'd definitely like to. |
| 23:34.48 | Kobaz | bcrisp: that's you wouldn't be able to use anything that's built in |
| 23:34.55 | Kobaz | bcrisp: but sure, you can do that |
| 23:35.08 | bcrisp | Kobaz that would be a neat feature |
| 23:35.22 | Kobaz | bcrisp: i don't know how you would retain the position in queue |
| 23:35.33 | bcrisp | i think southwest airlines does that |
| 23:36.09 | Kobaz | bcrisp: but when leaving a message, you can spawn an Originate() that will have one end join the queue, and one end do a Playback() of the message |
| 23:36.29 | Kobaz | dan__t: ael will make it easier for you to code, if you're a good coder or not |
| 23:36.32 | bcrisp | interesting |
| 23:36.42 | Kobaz | it's *much* easer to work with structured code than unstructured |
| 23:37.24 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 23:37.28 | Kobaz | bcrisp: all the tools are there, in asterisk... to do anything you could possibly want in a phone system |
| 23:37.35 | Kobaz | bcrisp: it's a matter of putting the pieces together |
| 23:37.46 | bcrisp | Kobaz: yep, i continually go over the applications list |
| 23:37.53 | bcrisp | need to learn AGI |
| 23:38.01 | Kobaz | agi is just stdin/stdout to asterisk |
| 23:38.02 | flux_control | I wish there were more softphones that supported iax(2)... |
| 23:38.09 | Kobaz | and running dialplan applications, getting variables, etc |
| 23:38.22 | p3nguin | Why more? You really only need one. |
| 23:38.25 | Kobaz | so you have the power of a real language, like c, perl, python, etc |
| 23:38.38 | Kobaz | flux_control: i can think of about a half a dozen |
| 23:38.48 | bcrisp | Kobaz yes i like :) |
| 23:39.01 | flux_control | Kobaz: For linux? Preferably CLI and/or curses (non-GUI)? |
| 23:39.04 | Kobaz | ael is not a real language |
| 23:39.11 | Kobaz | flux_control: non-gui, good luck |
| 23:39.37 | flux_control | I am only aware of about half a dozen *total*, and there seems to be more for Windows than linux. |
| 23:40.16 | flux_control | There are far more SIP clients than IAX clients overall. |
| 23:40.20 | Kobaz | but, for linux... zoper, iaxcomm, kiax, idefisk, and umm |
| 23:40.31 | Kobaz | there's some more |
| 23:40.38 | flux_control | I believe you meant zoiper. |
| 23:40.44 | Kobaz | that too |
| 23:40.50 | flux_control | Also, didn't idefisk get renamed? |
| 23:40.56 | p3nguin | You can't really include idefisk if you said zoiper. |
| 23:40.57 | Kobaz | probably |
| 23:41.05 | Qwell | idefisk ~= zoiper |
| 23:41.14 | Kobaz | kiax2 |
| 23:41.19 | flux_control | Ah, yes. |
| 23:41.20 | flux_control | :) |
| 23:41.38 | Kobaz | there's always libiax, and libncurses |
| 23:41.40 | Kobaz | have at it :P |
| 23:41.44 | flux_control | If I have time, I might. |
| 23:41.55 | flux_control | Right now the spare time department is rather lacking for me. :( |
| 23:41.58 | Kobaz | heh |
| 23:42.04 | Kobaz | lot of that going around |
| 23:42.08 | flux_control | At least there's asterisk console. :-D |
| 23:42.25 | Kobaz | yeah, i use asterisk as a test phone |
| 23:42.38 | Kobaz | it's a fully programmable soft phone, if you think about it |
| 23:43.00 | *** join/#asterisk ticoit (n=ticoit@201.191.190.123) |
| 23:43.11 | flux_control | Yeah, just seems to want full control of the soundcard under alsa. |
| 23:43.25 | dan__t | Kobaz, I was talking in regards to AGI, no AEL. |
| 23:43.37 | Kobaz | dan__t: about what |
| 23:43.52 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 23:43.55 | dan__t | AEL I've got down pretty well. I understand AGI well, but I'm not too good at making those AGI applications. |
| 23:44.18 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 23:44.38 | Kobaz | if you're not good at making agi applications, then most likly you don't fully understand agi |
| 23:44.56 | dan__t | I understand AGI. I promise. |
| 23:45.17 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:45.26 | flux_control | Is there a way in the dialplan to get a devices state before actually dialing the device? |
| 23:45.56 | Kobaz | flux_control: offhand i don't know the app/function |
| 23:45.59 | flux_control | s/devices/device's/ |
| 23:46.06 | Kobaz | flux_control: in theory, yes you can get any information... but! |
| 23:46.29 | Kobaz | flux_control: there is absolutly no guarantee that if the device is free when you check it, it will still be free when you dial it |
| 23:46.33 | flux_control | I know after calling Dial ${DEVSTATE} gets set, and I could use that. |
| 23:47.09 | flux_control | Kobaz: That's true. I was looking more for online vs. offline (which can still change in that amount of time). |
| 23:47.15 | Kobaz | chanisavail() |
| 23:47.30 | bcrisp | Kobaz: can I make * handle DTMF input while waiting in a queue? |
| 23:47.46 | bcrisp | i.e. caller A calls, is waiting with on hold music and types 123 for the voicemail option |
| 23:47.52 | flux_control | I was playing with music on hold, and I set it up to use music on hold as the ring (Dial(SIP/exten,20,m)), but in order to do that I need to Answer() first. |
| 23:47.59 | Kobaz | bcrisp: using res_features would work |
| 23:48.07 | bcrisp | Kobaz: thanks, ill look into it |
| 23:48.34 | Kobaz | flux_control: yes you need to answer in order to get out of the ringing state (ie: play tracks) |
| 23:48.48 | flux_control | If the callee is already offline before the Answer, then it doesn't make sense to do the Answer first |
| 23:49.07 | Kobaz | yeap |
| 23:49.20 | flux_control | So if there were a way to check state separately, I could check the state and only go to Answer if the callee is online. |
| 23:49.32 | flux_control | If the callee goes offline after that, it makes sense anyway. |
| 23:49.38 | flux_control | Well, to me at least :0 |
| 23:49.39 | flux_control | :) |
| 23:49.43 | Kobaz | chanisavail() on a sip/iax/dahdi line, will tell if if it's available |
| 23:49.49 | Kobaz | so if you have say, a sip phone |
| 23:49.55 | Kobaz | and it's unregistered, it will return unavailable |
| 23:50.06 | Kobaz | if it is registered (even if it's on a call), it will be available |
| 23:50.27 | flux_control | Ah |
| 23:50.42 | flux_control | Cool, that would work great, thanks! |
| 23:50.51 | Kobaz | there's some cool tricks you can do with chanisavail() to check for valid contexts and extensions |
| 23:51.01 | Kobaz | you can do chanisavail(local/exten@context) |
| 23:51.14 | Kobaz | and it will return whether or not that exten@context is in your dialplan |
| 23:51.32 | flux_control | I'm not too clear on what exactly the local channel is for. |
| 23:51.39 | Kobaz | The local channel is amazing |
| 23:51.44 | Katty | hi |
| 23:51.52 | Kobaz | It's a 'fake' device |
| 23:51.54 | flux_control | I just recently started playing with asterisk (although that's probably already evident) ;) |
| 23:51.55 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 23:52.06 | Kobaz | you can do anything to it that would be done to any other device |
| 23:52.31 | Kobaz | it really pollutes your cdr's though |
| 23:52.54 | flux_control | I currently don't have cdr's working (no logging is working for me). |
| 23:52.57 | Kobaz | heh |
| 23:53.04 | flux_control | Hopefully that will get fixed via upgrade. |
| 23:53.20 | dan__t | I just got pegged in the side of the head by an airsoft pellet. |
| 23:53.34 | Kobaz | fun |
| 23:53.34 | dan__t | Work decided it would be a great idea to give us all airsoft guns as a Christmas present. |
| 23:53.47 | Kobaz | i should get my employee a christmas present |
| 23:55.24 | *** join/#asterisk Benny_132 (n=benny_13@59.167.161.153) |
| 23:55.55 | flux_control | I just looked on asterisk.org/docs about ChanIsAvail, and it lists the different variables that get set, but doesn't have a list of what values they can contain.. :( |
| 23:56.49 | Kobaz | much of the ast docs are lacking |
| 23:57.38 | Kobaz | AVAILSTATUS is a 0 or 1 |
| 23:58.26 | bcrisp | im not seeing res_features.so in the list of modules and didnt see it in * make menuselect |
| 23:58.38 | Kobaz | bcrisp: it's built in now |
| 23:58.44 | bcrisp | Kobaz: phew |
| 23:58.49 | Kobaz | it was a module |
| 23:59.10 | bcrisp | neat thanks |
| 23:59.30 | Kobaz | bcrisp: i believe you're going to have to do some trickery to get features to actually work when you're in a queue |
| 23:59.46 | Kobaz | because I think they are only available when you Dial() |
| 23:59.50 | bcrisp | ah |
