IRC log for #asterisk on 20141128

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01:26.13MilosWhat does "Peer is not supposed to register" mean?
01:26.35WIMPyIt's not dynamic.
01:51.11MilosIs it possible to restrict asterisk debug to just a single host?
01:55.18WIMPyFor sip? Yes. 'sip set debug ip <ip>'
01:55.46[TK]D-Fender<PROTECTED>
01:56.18[TK]D-Fender(if it has an known host)
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02:30.20MilosHow do I disable registration when the other end is set up as host=<static IP>?
02:30.30MilosI don't have an explicit register => line
02:31.43[TK]D-FenderMilos, the OTHER end is trying to register TO Asterisk
02:32.12[TK]D-FenderMilos, Make that other side stop trying to register
02:33.55MilosI'll just confirm so that there is no misunderstanding - I have server A and server B that are both asterisk. server A is the ITSP and server B (me) is trying to set up a trunk to that server.
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02:34.03Miloswhen you say "other side" which side are you talking about?
02:34.28[TK]D-FenderOne of these is trying to register to the other
02:34.36[TK]D-FenderNot sure how much clearer one can be...
02:34.59MilosServer B (me) seems like it is trying to register, but I have no explicit register line in my config.
02:35.14[TK]D-FenderAnd the one that is doing the registering is trying to do this to a server that has a peer that has a fixed host specified
02:35.18Milosjust type=friend,context=blah,host=serverB,insecure=port,invite
02:35.24[TK]D-Fenderit is therefor not ALLOWED to register at all
02:35.49[TK]D-FenderYou if you put an IP then they will not be permitted to register.
02:36.01Milosyes so going back to my question
02:36.05Milos<Milos> How do I disable registration when the other end is set up as host=<static IP>?
02:36.10[TK]D-Fender...
02:36.14[TK]D-Fenderfix the side that is doing it
02:36.24Milosthat's what I'm asking about
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02:36.29MilosI'm asking why my side is trying to register
02:36.31MilosI never told it to
02:36.33[TK]D-Fender....
02:36.38[TK]D-Fendershow us..
02:36.48Miloswhat would you like to see?
02:36.55MilosI have no "register =>" line
02:36.57[TK]D-FenderSIP DEBUG from CIL
02:37.14[TK]D-Fenderbecause your description is starting feel very vague as to which system is which
02:37.15WIMPyAre you sure it's your server that's trying to register?
02:37.23MilosI'll provide debug
02:37.34[TK]D-Fender~pb
02:37.34infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
02:37.35[TK]D-Fender^^^
02:41.39Milos[TK]D-Fender, http://dpaste.com/0QSHJ1Z
02:41.41Milosis that enough output?
02:42.09Milosthe other end is saying [2014-11-28 15:42:01] ERROR[27945]: chan_sip.c:15436 register_verify: Peer 'johnmilos' is trying to register, but not configured as host=dynamic
02:42.10[TK]D-FenderReliably Transmitting (NAT) to 113.21.227.45:5060:
02:42.10[TK]D-FenderREGISTER sip:jarvis.hd.net.nz SIP/2.0
02:42.18[TK]D-FenderYup, you are indeed trying to register
02:42.21Milos<Milos> How do I disable registration when the other end is set up as host=<static IP>?
02:42.27MilosI've asked 3 times
02:42.38[TK]D-Fenderit isn't normally part of the peer
02:42.41[TK]D-Fendershow us your config
02:43.09Milos[TK]D-Fender, http://dpaste.com/35CE57Z
02:43.15WIMPyRemove it from your config and restart chan_sip or the whole Asterisk.
02:43.22[TK]D-FenderMilos, ...
02:43.26MilosI bet you when I restart all of asterisk it will stop registering.
02:43.28[TK]D-FenderREGISTER is not part of the peer
02:43.32Milosyes it is
02:43.33[TK]D-Fenderyou are restricting your view
02:43.39MilosI don't use [general] register => lines
02:43.49MilosI use callbackexten
02:43.50[TK]D-FenderYou have had a register command in the main config under [general]
02:43.56Milosno I do not
02:44.01WIMPyA simple 'sip reload' is not enough to remove registrations as far as I experienced it.
02:44.05Milos^
02:45.01Milosproblem solved.
02:45.32[TK]D-Fender; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
02:45.32[TK]D-Fender747 ; this is equivalent to having the following line in the general section:
02:45.32[TK]D-Fender748 ;
02:45.32[TK]D-Fender749 ; register => username:secret@host/callbackextension
02:45.49[TK]D-FenderSo putting that is is the same as a register outside
02:46.13Milosyeah
02:46.15Milosbut I removed it
02:46.21Milosand sip reload doesn't remove it
02:46.21[TK]D-Fenderif it gets stuck after you've changed your config you should be able to pull it out of astdb
02:46.27Miloshow?
02:46.38[TK]D-Fenderdatabase <tab>
02:46.47[TK]D-Fenderdo a show them go find the key for it
02:47.38WIMPyWhat do you see there? I have never seen anything related to outbound registrations in AstDB.
02:48.13[TK]D-Fender<PROTECTED>
02:48.42WIMPyThat's inbound.
02:49.39[TK]D-Fendernope
02:49.50[TK]D-Fenderumm...
02:49.55[TK]D-Fenderhold that thought
02:50.03[TK]D-Fenderlooking like
02:50.11[TK]D-FenderGuess it may jsut be cached in ram
02:50.15[TK]D-Fendermight need that restart
02:50.53WIMPy'module unload chan_sip' and 'module load chan_sip'.
02:51.02[TK]D-Fenderthat'd do it
02:53.33Milosthat's a little unfortunate
02:53.38Milosbut I guess it's not a common requirement
02:53.45Miloswould that kill active calls?
02:54.31Penguin
02:55.24[TK]D-Fenderyup
02:55.37WIMPyYou probably can't do it with active calls.
02:55.52[TK]D-FenderYou could... if you can live with them dropping :)
02:55.58WIMPyNope. Not possible.
02:56.02[TK]D-Fender"core restart when convenient"
02:56.22WIMPyYes. That's the way.
02:56.41Penguincore restart gracefully
02:57.22WIMPyIf you want to prevent users from making calls, yes.
02:57.38PenguinEventually, no more calls will exist.
02:57.51PenguinAnd it will restart.
02:58.03WIMPyYes, but it could mean some hours of downtime.
02:58.39PenguinIf you have single calls lasting hours, you're right.
02:58.55PenguinIf someone hangs up a call, they can't make another one until after the restart.
02:59.10Miloswow that's an option?
02:59.16Milosis that valid syntax "core restart when convenient" ?
02:59.19PenguinSo unless there are calls remaining for hours, it's restart much sooner.
02:59.20WIMPyHappens far too often.
02:59.32WIMPyMilos: yes
02:59.32PenguinYes, that's the right syntax.
02:59.36PenguinLearn to use your tab key.
02:59.41Penguincore restart TABKEY
03:00.31Milosawesome
03:00.40Penguincore restart when convenient will restart as soon as no channels still are open, but new channels will be allowed to be started up.
03:01.10Penguincore restart gracefully will restart as soon as no channels are open, but it prevents new calls from happening.
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03:05.18[TK]D-FenderBasically leads to increased user frustration until everyone gives up of their own volition ;)
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04:06.00Kobazoh
04:06.01Kobazyeah
04:06.28Kobazi've had problems with sip peers getting stuck. and yeah restart, or module unload/load is the one way to fix it
04:06.37Kobazthe other way is to delete the peer, reload, and then put the peer back
04:06.56Kobaza bit safer and no downtime
04:08.13WIMPyIt was about the inability to remove it.
04:10.20Kobazmm
04:10.41Kobazyeah, that i've never run into, but my point was i've seen it elsewhere in the sip stack
04:11.54WIMPySo what kind of stuck peer did you experience?
04:12.08Kobazso, i had a peer set to say host=1.2.3.4
04:12.22Kobazand then you edit the peer and do host=4.5.6.7
04:12.23Kobazsip reload
04:12.29Kobazand it's still at 1.2.3.4
04:13.04Kobazif you change the name of the peer, sip reload, and then change it back, then the address will update
04:13.08Kobazit's really rare though
04:13.18Kobazi can't reproduce it consistently
04:14.22WIMPyI can't remember details, but I also experienced that changing a peer doesn't always work.
04:15.06WIMPyLuckily not something I do normally.
04:15.08Kobazyeah
04:15.33Kobazhttp://9gag.com/gag/a3d183v?ref=fsidebar
04:15.36Kobazwhoops
04:15.39Kobazwrong channel
04:15.41Kobazbut you can look anyway
04:23.34PenguinThe simple answer: her.
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04:26.27Kobazheh
04:26.56WIMPyWould you dare to look what's beneath the horse head?
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04:30.06PenguinLeave it on or turn off the lights.
04:41.55Kobazchromecast is pretty cool
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08:56.40PenguinI finally got my new asterisk box in place.
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10:27.41casdude<PROTECTED>
10:27.49casdudemy current setup is [telco BT ISDN30] > [asterisk 1.8.13.1] > [PBX Samsung 7200]
10:28.11casdudeWhen receiving calls the systems work fine, however outbound calls do not work. I receive Accepting overlap call from '3207' to '<unspecified>' on channel 0/1, span 2, and then the call is hungup.
10:29.01casdudei thought it would be dial plan but it looks ok http://pastebin.com/TiEiuhUr
10:29.22casdudeit just appears that the samsung is not sending the numbers to be dialed
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10:39.22casdudei thought it might be indictation but again these look ok to me based on the uk http://pastebin.com/LBvaBMLb
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11:37.12WIMPycasdude: That's a pretty interesting dialplan you have there.
11:39.34WIMPyI can't remember what way I got it working with dahdi, wehn I tried that. IIRC it was with immediate=yes and WaitExten.
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11:40.17WIMPyIndications don't mean anything. If you want to show something, show us a call.
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11:59.04casdudesorry wimpy took a call will see if i can grab hold of one now
12:00.30casdudea call is fairly brief tbh http://pastebin.com/EL4zYQTX
12:02.21casdudeattemps to make the call and then hangs it up
12:03.09WIMPyBecause you don't have an s extension.
12:06.42casdudewe dont have any extension the call is passed through from span 2 to span 1 [PBX Samsung 7200] > [asterisk 1.8.13.1] > [telco BT ISDN30]
12:09.01casdudeI would have thought the number would still be present, currently it states it as "Accepting overlap call from '3207' to '<unspecified>' on channel 0/1, span 2" the <unspecified> is where the issue is i think
12:09.38WIMPyThat's just the way it is. You need some dialplan to forward the call.
12:12.01casdudesure, now you come to say it, it does seem strange, let me check some things. Thanks for your time
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16:46.50tzicaI have setup a trunk between asterisk/freepbx and a 3rd party  and receiving 401 Unauthorized when calling from 3rd party to Asterisk
16:46.53tzicaany advice ?
16:49.00[TK]D-Fenderanswered in #freepbx
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17:25.44jcongdonHello, is there an easy way to stop a dial?  I use the manager port to control calls.  I make my caller connect to another person, but may want to stop that mid dial if possible.
17:26.13WIMPyHow? And what do you want to happen then?
17:26.54jcongdonWe have people dial into our system.  Then when the click a button on our web interface, they will dial a contact.
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17:27.05jcongdonIt may start dialing, and then they want to stop that connection
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17:27.13jcongdonTrying to figure out the best way to stop it
17:27.38WIMPyJust send a Hangup.
17:28.44jcongdonOk, that's where I'm having trouble.  I don't want to hangup on the member connected to me.  I guess I need to find the channel that hte dial command is going out on
17:29.32WIMPyYou can also 'Redirect' the channel to some (other) place in your dialplan.
17:29.58jcongdonThat will cause the dial to stop?  I can do that
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17:30.10jcongdonThanks WIMPy
17:30.17WIMPyIt will end the running App, yes.
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18:53.48Kobazi like to Redirect to random locations
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19:34.05dimitry7Hi guys. Sorry I am trying to compile asteriks
19:34.15dimitry7but I do need the cdr_pgsql module
19:34.38dimitry7I have unmet dependencies, but I cant see which ones, how may I see them or what are they??
19:38.21dimitry7hello
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20:01.20ChannelZyou probably need the pgsql devel stuff
20:02.52ChannelZpgsql-devel or libpostgres-devel or somesuch
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20:08.39dimitry7ChannelZ, ohhh
20:08.41dimitry7great, let me try
20:12.51dimitry7ChannelZ, GREAT IT WORKED
20:12.59dimitry7my man... thank you so much!!!! :D
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21:07.05ChannelZno problem
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23:17.30Kobazsooo
23:17.53Kobazif someone does a 302 moved temporarily
23:17.59Kobazasterisk makes a happy little local channel
23:18.25Kobazis there a setting to make that local channel be optimized out as early as possible?
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23:47.39Kobazi think i can fix it by doing ast_set_flag(c, LOCAL_BRIDGE); ast_clear_flag(c, LOCAL_NO_OPTIMIZATION);   in app_dial do_forward()

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