IRC log for #asterisk on 20151231

00:01.43ModFather[TK]D-Fender how are you doing?
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06:22.51game0guys, I'm recieving this error while opening elastic:  acl.c:962 ast_ouraddrfor: Cannot connect
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06:49.10game0I have configured the hostname for my machine /etc/hosts
06:49.27game0but still not sending this error why it's like that
06:49.28game0?
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08:26.14game0guys, I'm recieving this error while opening elastic:  acl.c:962 ast_ouraddrfor: Cannot connect
08:26.18game0I have configured the hostname for my machine /etc/hosts
08:26.24game0but still not sending this error why it's like that
08:26.26game0?
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10:01.25WIMPyOh cool. The SIP attacks are back.
10:23.40game0hi WIMPy, how are you?
10:25.52WIMPyTired
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10:31.36game0ok
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13:17.28stkochhi, is this possible in a dialplan: exten => _["0"|"0049"|"+49"]1[567].,1,Dial(...)    ?
13:17.58WIMPyno
13:18.16stkochis it otherwise possible with one line?
13:18.17WIMPyYou have to make multiple extensions.
13:19.54WIMPyIt is however possible to canonicialise the numbers with the help of some loopback switches. Or just Goto if you never dial interactively.
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13:21.42BommyHi
13:22.09BommyI'm looking for assistance with sending faxes
13:22.28BommyI have Elastix installed, and faxes are being sent using HylaFAX
13:23.10BommyThe problem I have is with the conversion of pdf to fax, using ghostscript. the result quality is very bad.
13:24.00BommyI am looking for a way to convert it to Group 3 or Group 4 tiff, to be able to send it through the fax, but in better quality than the default conversion
13:24.08Bommydid anyone have experience with such thing?
13:27.46WIMPyMake sure you don't output greyscales and then try to optimize other parameters.
13:28.54BommyThis is the command that runs by the HylaFax pdf conversion:
13:28.55Bommygs -q -sDEVICE=tiffg3 -dNOPAUSE -dSAFER=true -sPAPERSIZE=a4 -dFIXEDMEDIA -dMaxStripSize=0 -dBATCH -r209.10x196 -sOutputFile=fax.pdf.out fax.pdf
13:29.23Bommywhen I replace to tiffgray it doesn't work, because it doesn't have compression of group 3 or group 4, and therefore doesn't send the fax.
13:29.36Bommywhen using the original, its output sucks
13:29.47BommyI tried using also ImageMagick for the conversion, with no much success.
13:30.46Bommythe "best" result I had with the conversion is when I converted with ghostscript to tiffgray, and then with imagemagick to add compression. the file looked better in my preview, but when sending it via the fax, it inverted the black/white colors for some reason
13:32.24stkochWIMPy: is it possible to have an extension with exten => _00^(49).,1,Dial(...) ? This should mean 0041, 0043 but not 0049
13:40.43stkochor how to disable automatically fallback to next extension example:
13:41.16stkochexten => _00491[567].,1,Dial...
13:41.58stkochexten => _0049.,1,Dial(second provider...)
13:42.21stkochif the first is not available, then automatically the next extension is used
13:42.34stkochthis should be disabled - is it possible?
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13:48.58WIMPystkoch: Always End your extensions with an explicit Hangup() unless you want them to fall tthrough to other matching extensions.
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14:00.02DovidHi. The > indicates what I am getting from the far end or what I am sending out ? http://pastebin.com/MjVR99x2
14:00.58WIMPyIt's reverse.
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14:01.12WIMPyI.e. < is Rx and > is Tx.
14:01.26DovidWIMPy: Thanks. its what I assumed.
14:01.39Dovidcarrier insists the PRI is up. All I see is > and I wanted to make sure
14:01.45Dovidtime for a loop test
14:02.10WIMPyWhat abou L1 state?
14:02.15WIMPy+t
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14:06.32stkochIs it possible to use AST_CONTROL_FLASH instead of *2 for attended transfer ?
14:06.47WIMPyWhere?
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14:13.32stkochWIMPy: in dialplan
14:13.44stkochWIMPy: sorry
14:14.03WIMPyYou don't do transfers from the dialplan.
14:14.34WIMPyAnd if you come up with those variables, that looks like you're fiddling around with the source code.
14:14.59stkochWIMPy: I mean, if I call and then want to transfer I could press *2 (features.conf) and transfer the call - this works
14:15.41WIMPyThe way you can do a transfer depends on what kind of device you're using.
14:15.43stkochbut POTS telephones have an hook flash button
14:16.15stkochSo I mean, press the hook flash button should do the same as *2
14:16.29WIMPyAnd DTMF features is the possibility that works wherever DTMF works.
14:16.41WIMPySo are you using a POTS phone?
14:17.01stkochyes with chan_lantiq
14:17.22WIMPyWhat the heck is that?
14:17.35stkochThe disadvantage of DTMF is that the callee hears *2 but not hook flash
14:17.58WIMPyIt should get filtered.
14:18.12stkochWIMPy: https://dev.openwrt.org/browser/packages/net/asterisk-1.8.x/src-lantiq/channels/chan_lantiq.c?rev=33995
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14:18.51WIMPyCan't connect there. So it's for som IAD FXS port?
14:19.17WIMPyIt then depends on what chan_lantiq offers.
14:19.18stkochWIMPy: yes for Lantiq based routers
14:19.28stkochI have patched chan_lantiq:
14:19.30stkochast_queue_control(iflist[i].owner, AST_CONTROL_FLASH);
14:19.55stkochIf the hook flash is pressed, the control flash is recognized by asterisk
14:20.16stkochBut where to handle the control_flash ?
14:20.18WIMPyThe whole hold and opening of a 2nd channel would have to be done there.
14:20.34WIMPyYou can steal from chan_dahdi.
14:21.25stkochWIMPy: so it is not just possible to interprete AST_CONTROL_FLASH like *2 from features part ?
14:21.39WIMPyNot as far as I know.
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14:22.36stkochWIMPy: thanks
14:42.27stkochWIMPy: is this valid: _[X+]
14:42.42stkochI want to allow 0 1 2 3 4 5 6 7 8 9 and +
14:42.55WIMPyYes. But it probaly makes no sense.
14:43.07WIMPyOnly one character?
14:43.19WIMPyThat's what it does.
14:44.24WIMPyErr. Garbage. X only works outside []. You use [] to escape a literal X.
14:44.31WIMPySo you need [0-9+].
14:46.18stkochWIMPy: because I have read that _. should not used because it allow other like t and so on
14:46.41WIMPyThat is true.
14:47.50stkochneeds plus an escape sign? or what would be in this case [0-9-] if I want allow the minus
14:49.04WIMPyUgh. Can't remember. I think you put - as the 1st char.
14:50.49stkochif I use _[+0-9]. then I can enter at minimum 2 digits, could be the . optional in one line?
14:51.22WIMPy. means one or more characters.
14:51.35stkochwhat is with zero or more chars ?
14:51.48WIMPy!
14:53.07WIMPyThere is however another difference betwenn . and ! in that ! will match as soon as nothing else could possibly match while . will wait for a timeout.
14:57.29stkoch_[+0-9]. does not work for one digit
14:57.35stkoch_[+0-9]! does work for one digit
14:57.55WIMPyThat's what I said.
14:58.03dan_jHi. Does anyone know what the maximum length is for a mailbox number?
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14:58.29WIMPydan_j: Depends on Asterisk version.
14:58.39dan_jv11 at the moment
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14:59.22WIMPyI don't know the limit, but I can fit a complete phone number there now without issues.
15:00.00dan_jok. cool. I'll give it a try. Thanks.
15:01.03WIMPyOh, and it's a name, not a number.
15:04.10dan_jThe field called mailbox in realtime
15:05.01WIMPyThat should be the name.
15:05.23dan_jOk. It's that field that i'm thinking about.
15:05.32dan_jRegarding name, there is a field called fullname
15:06.03WIMPyThe owners name.
15:18.07dan_jOk. Thanks.
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15:23.07stkochCan I add an argument to Goto() ? And get that with ${ARG1}
15:23.27WIMPyNo. Only Gosub.
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15:59.14steelbrainI'm getting this error after following the PJSIP realtime tutorial, any ideas?
15:59.14steelbrain[Dec 31 15:58:21] WARNING[3994]: sorcery.c:997 __ast_sorcery_insert_wizard_mapping: Wizard 'realtime' failed to open mapping for object type 'endpoint' with data: ps_endpoints
15:59.15steelbrain[Dec 31 15:58:21] ERROR[3994]: sorcery.c:775 __ast_sorcery_open: Error attempting to apply configuration res_pjsip to sorcery.
15:59.15steelbrain[Dec 31 15:58:21] ERROR[3994]: res_pjsip/config_system.c:157 ast_sip_initialize_system: Failed to open SIP system sorcery
15:59.16steelbrain[Dec 31 15:58:21] ERROR[3994]: res_pjsip.c:3875 load_module: Failed to initialize SIP 'system' configuration section. Aborting load
15:59.22steelbrainTutorial Link: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
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16:18.23steelbrainI fixed via a clean reinstall
16:21.23steelbrain----------------------------------------------------
16:21.35steelbrainCan I recommend a wiki correction? I don't have a login and there's a typo that cost me a lot of time
16:21.54steelbrainres_odbc.conf <-- in this file, the section of the name should be `asterisk-realtime` instead of `asterisk`
16:21.59steelbrainTutorial Link: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
16:22.47steelbrainThanks guys, Happy New Year!
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16:57.44newtonrDO you mean the DSN or the context/section name?
16:57.47newtonroh he is gone
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23:16.31pjensen00Question.  When there are three channels in the same bridge, and one channel is put on hold is it expected that hold music is played back to the other two participants in that bridge?
23:17.16WIMPyWhat exactely is put on hold?
23:20.36pjensen00I am pushing the physical hold button, so the SIP invite sent from my phone to asterisk
23:21.07pjensen00When I have just two people in a bridge, the hold music plays.  Three or more in the bridge and no music
23:22.23WIMPyWould be quite annoying if everybody could disturb a conference by putting it on hold, wouldn't it?
23:22.36pjensen00It would!
23:22.59WIMPyAlthough that is exactely what's known to happen if you connect from Asterisk to Asterisk via SIP.
23:23.07pjensen00Though just earlier today someone did that to a conference bridge of ours, but since the call came from the PSTN we didn't get the SIP message indicating a hold
23:23.16pjensen00so we just got hold music to all other parties in the conference bridge
23:23.35WIMPyTe joys of using SIP.
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23:23.42WIMPyThe
23:23.43pjensen00I'm in the "what's supposed to happen" phase
23:23.47pjensen00Ha yeah.
23:24.16WIMPyWell, in theory it could work correctely. But as we know it's rather unlikely in practice.
23:24.37pjensen00So "correctly" is to not play music to 2+ party members?
23:24.52WIMPyIIRC there might be somethign about PJSIP <> PJSIP eliminating that issue.
23:24.59WIMPyOff course.
23:25.28WIMPyBut you need to know that you're on hold. So when that information is lost, you're screwed.
23:25.32pjensen00I'm trying to look up documentation at the moment to see if there's any official word.
23:25.38pjensen00Yeah, that was my gut feeling.
23:26.08WIMPyYo usually won't find that level of information.
23:26.17pjensen00:( curses.
23:26.44WIMPyThat's what you'd expect in Specifications. We only hace documentation to some degree.
23:28.41pjensen00ah, well thanks for the insight.
23:44.01tompawHello.
23:44.13tompawDid anyone experiment with ari & openstack?
23:49.16pjensen00Just ARI.  :3

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