IRC log for #asterisk on 20140911

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07:00.23flingWhich service is better for outgoing calls to skype?
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07:34.18ChannelZWhat are my choices?
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09:13.27adscwhat is the most frequent problem when a registered SIP phone can't dial out?
09:13.51adscmy asterisk CLI with verbosity 10 shows nothing when I try to dial out from this phone
09:14.01adscdialing in works fine
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09:18.10adschere's a paste with my configuration: http://pastie.org/9544377
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09:44.41adscshouldn't I at least see something in the console?
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09:47.04kleszczyes
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09:50.22kleszczset verbose to 7 and debug to 2 and check again
09:54.22adscnothing
09:54.30adsci am beginning to suspect the hardphone
10:06.42adscyeah, the hard phone was set up to use fixed line for outgoing
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10:28.23wasanzyhello friends
10:28.54wasanzyplease how can I configure asterisk to accept incoming calls from anywhere?
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10:55.09wasanzyany help please?
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11:06.56flingWhich modem to buy for a gsm gateway?
11:07.12eirirs_us robotics 56k fax modem
11:07.13flingdo I still need e1550 or is it possible to use something more recent?
11:07.15flingLOL
11:07.31flingeirirs_: I have this one but it got damaged by lighting one day
11:07.36eirirs_:(
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11:33.32wasanzyplease how can I configure asterisk to accept incoming calls from anywhere?
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13:10.45funnymanvaDoes anyone have experience with Digium's PSTN GW products like G100, G200, etc?  I'm trying to figure out how to set the codec used on the E1 interface (aLaw or uLaw) and I don't see the settings anywhere.
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13:41.31malcolmde1 should default to alaw.  you shouldn't have to set it.
13:44.31WIMPyHmm. Pretty interesting. The only thing you don't have to explicitely configure manually?
13:52.32funnymanvamalcomd thanks, we have really horrible audio so I was hoping it was a codec issue, but maybe it's an encoding issue.  They use B8ZS, but it doesn't appear the G400 can support that.
13:56.38[TK]D-FenderIt does
13:56.44[TK]D-Fenderthat is absolutely bog standard
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13:59.32funnymanvaFor Coding on the E1 it only has AMI and HDB3 as options, I see no selection for B8ZS on the G400
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14:12.18malcolmdthe e1 equivalent of t1's b8zs is hdb3.  you've got a carrier that's requiring/requesting b8zs on an e1 link?
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14:13.55funnymanvamalcomd, we're interfacing to a Nortel PBX (not under our control) and that's what the guy was telling us.  I don't think it's an encoding issue or signaling wouldn't be working.  I think they're using ulaw on E1 trunks (this they did confirm) instead of alaw which is why the audio sounds horrible (almost unintelligible like we're inside an airplane engine).
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14:16.42malcolmdphone up the Support guys
14:18.15malcolmdand make sure you're setting the right value for Use Internal Timing on the T1/E1 General Settings configuration menu
14:19.00funnymanvaok, thanks
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14:59.53marceloamorimguys, morning, I`m trying to use the calendar.conf, and there is an error, I don`t know if I`m doing the right way. when I use the CLI> calendar show calendar mygooglcall all the informations is there
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15:00.39marceloamorimbut when I set NoOp(${CALENDAR_EVENT(summary)}, the error: There is no event notification datastore on 'SIP/3000-00000001'!
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15:05.32marceloamorimshould I query the agenda first?
15:14.05marceloamorimTo get that, I need to query the calendar first. Very useful. Thx guys
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15:53.01Stefan27Does the sound prompt searching as in https://wiki.asterisk.org/wiki/display/AST/Sound+Prompt+Searching+based+on+Channel+Language really work as the wiki page says? I tried putting Set(CHANNEL(language)=en_test) and then Playback(mooon.wav) but this only works when i have created a file called mooon.wav in the sounds/en directory, not when i create it in the sounds/en/test directory
15:53.34Stefan27<PROTECTED>
15:53.34Stefan27<PROTECTED>
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15:53.51Stefan27works when moon.wav exists directly in en but not when it only exists in en/test
15:53.55[TK]D-Fender"not when i create it in the sounds/en/test directory" != "en_test"
15:54.19Stefan27How does en_test work then?
15:54.35[TK]D-Fender<PROTECTED>
15:54.49[TK]D-Fenderthat underscore doe not magically become a slash
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15:55.01Stefan27I thought it would, according to the wiki page i linked
15:55.13Stefan27Each channel in Asterisk can be assigned a language by the channel driver. The channel's language code is split, piece by piece (separated by underscores), and used to build paths to look for sound prompts. Asterisk then uses the first file that is found.
15:55.38[TK]D-Fendernot to my awareness
15:55.53[TK]D-Fenderuse core debug to confirm
16:01.47Stefan27core debug doesn't tell me anythign useful, but it does locate the file in en/ with language set to en_test so it must parse it somehow (there's no folder called en_test nor any file called mooon.wav anywhere but in en/
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16:10.00WIMPyThe versions remain open. Both for the wiki and what you use.
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16:10.44Stefan27I use 12.5
16:11.03Stefan27wiki was updated dec 19 2013 thats all i know
16:11.34WIMPyIf the date is correct, that must be new enough for whatever is documented in the wiki.
16:13.51mjordanThat code hasn't changed from god only knows when.
16:14.00mjordanI'm not sure the wiki page is correct however.
16:14.20mjordanAsterisk will do a fallback, in that if you have a sound in en_foo, it will play it - but if that sound doesn't exist, it will check en next
16:14.49mjordanWhat I'm not sure of is that it will form a path search from the underscores.
16:15.02mjordanBut I haven't dug deep enough to know for sure.
16:17.33mjordanlooking all the way back to 1.8, it looks like it would not form a sub-directory search. I think the wiki page is probably just wrong.
16:17.48mjordanBased on the example there, it would instead do the following:
16:18.22mjordanIf the channel language is "en_GB_female_BT", check for that folder first, i.e., sounds/en_GB_female_BT
16:18.37mjordanIf it didn't find the sound, check sounds/en_GB_female
16:18.45mjordanagain, if it didn't find it, check sounds/en_GB
16:18.49mjordanand finally, check sounds/en
16:19.24mjordannewtonr: mind making an issue to fix that wiki page? https://wiki.asterisk.org/wiki/display/AST/Sound+Prompt+Searching+based+on+Channel+Language
16:21.55newtonrfor sure
16:22.02newtonror, not at all
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16:34.38newtonrhttps://issues.asterisk.org/jira/browse/ASTERISK-24319 created
16:46.26Stefan27setting language to 'en/test' made it behave almost like i wanted but not quite http://pastebin.com/XKbKseWS
16:47.01Stefan27I guess I have to re-make my own sound-file-structure based on how it actually works, thanks for telling me jordan
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16:49.19WIMPyYou might want to add that to the issue newtonr just created.
16:52.43Stefan27ok i added a comment
17:12.56DovidIs there any way of sending CALLERID(name) as blank? If I Do Set(CALLERID(name)=) then it will be the value of CALLERID(num) which I do NOT want.
17:14.27[TK]D-FenderShow us
17:22.41newtonrit may be you want to set CALLERID(all) as blank
17:22.54newtonrIIRC
17:23.57newtonralthough depending on the path of the call, if its going out of PSTN the provider may overwrite whatever you do
17:24.02newtonr*out over
17:27.21WIMPyApropos callerid: I'd still like to know what exactly CALLERID(ani) is supposed to be.
17:28.38WIMPyFrom the description, I expected it to be the cumber of the calling line (as opposed to the callers number), but that doesn't seem to be the case.
17:29.05WIMPyAnd if it isn't supposed to be that way, where would we put the calling lines ID?
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17:49.03Penguindovid: Does it have to be a null value?  Can you set it to UNKNOWN instead?
17:52.17PenguinIf I remember right, when I set my CALLERID(name) to UNKNOWN, I discovered an undocumented feature of my ITSP.  It actually caused my outgoing callerid info to be anonymous/restricted.
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18:01.35DovidPenguin: I can't. I need to pass the number but not the name. Some carriers (verizon fios and cablevision I believe) if they get something in the cnam field they wont do a cnam dip (to save money). As communicaitons in the US turns over to SIP more and more what you set will go end to end
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18:23.25Dovidin 11.4 if you do Set(CALLERID(name)=) then it wont sent anything
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18:23.44Dovidi wonder what changed. and where in the code can i look for this
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18:28.22[TK]D-FenderDovid: Show us....
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18:28.53Dovid[TK]D-Fender: A copy of both SIP traces?
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18:29.23[TK]D-Fender* CLI verbose 10, SIP debug.
18:29.39[TK]D-FenderAND the pee config
18:29.44[TK]D-Fenderpeer*
18:30.43Dovidstandby
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18:46.18Penguinnewtonr, mjordan, file, malcolmd
18:46.33filehm?
18:47.23Penguink5673 could use some ban time.
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18:48.30Dovid[TK]D-Fender: Here are the traces: http://pastebin.com/HGGfXNW8
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18:50.35[TK]D-FenderDovid: There is no name.... looks normal
18:50.37Dovidwith config here" http://pastebin.com/smf1Brzf
18:50.44[TK]D-FenderDovid: now go SET one and test
18:50.47DovidTK: The first one has no name, the second trace does
18:51.28DovidMy issue is that in ast 1.8.22.0 if I set Set(CALLERID(name)=) it will be set to ${CALLERID(num)}
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18:52.21[TK]D-Fender1.8.22 is 8 revisions out of date
18:52.21WIMPyTime to get rid of 1.8 then.
18:52.26[TK]D-Fender^
18:52.28[TK]D-FenderAnd that
18:52.40Penguinfile: How about a little /mode +b $a:k5673
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18:53.15DovidI know. I assume I can have it patched to act the way I need it?
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18:53.40DovidI stick with what works. I woild love to just move but when we find something stable we stay with it
18:53.50WIMPyThat (and a lot of other goodies) is called Asterisk 11.
18:54.11WIMPyYou just told us it doesn't work for you.
18:54.19DovidWIMPy: It's a minor issue.
18:54.33Dovidvs. much larger issues we have had in the past which makes us hesitant to mvoe
18:55.45mjordanDovid: which issues in particular have you had with 11
18:55.50mjordanAs in, which JIRA issues.
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18:56.03mjordanI'm honestly very curious, as the differences between 1.8 and 11 are not gigantic.
18:56.20mjordanAnd, where there were some issues, have mostly been addressed. The vast majority of the time when an issue is filed against 11, it also applies equally to 1.8.
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18:58.09PenguinI guess that'll work just as well.
18:58.25Dovidmjordan: I did not even touch 11 (for my regular job as opposed to my own stuff which is why I am mostly here). We have had many random issues, even in the same version such as in 1.8. in some versions > 1.8.22.0 Asterisk would random;y "forget calls". SIP calls would just disapear. When the RTP stops and the carrier sent a BYE asterisk would just send a 481. So we rolled back to 1.8.22.0. I dont remember them
18:58.26Dovid<PROTECTED>
18:58.57mjordan1.2/1.4 is out of my hands (and everyone around here for that matter). To be frank, that's like blaming Redhat for Fedora 4.
18:59.22mjordanFor 1.8, I'd really like to know what issues those were and have them reported. We take regressions pretty seriously, and try to make sure that when one happens, we get tests written to keep it from happening again
18:59.33mjordanIf there aren't specific issues in JIRA however, there's not much we can do
18:59.56Dovidmjordan: You are right and I am not blaming you for the issues that were there then. but becase of the "scary unknown" we stay where it works. One of these years I am going to load up 11 on a test box and see how it works
19:00.05Dovidfor my side work I am on 11 and its been great
19:00.26mjordanthat's good to hear; if you do find something please do let us know
19:00.26mjordan:-)
19:00.46Dovidmatt: I will. geez. so many people helping me here. i am gona owe a lot of beer
19:00.51DovidTK: Coming to Astricon?
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19:01.37[TK]D-FenderDovid: Nope.
19:03.32DovidTK: I dont think you ever came (I think I asked you every year....). it's a lot of fun. especially drunk Matt Riddell
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19:09.55QwellDovid: Matt Riddell is the best
19:11.23DovidQwell: Last year he didnt drink though. lets hope this year he is again. no offense to Digium or anyone else but last years party as a tad lame. the year before was fun
19:11.35Dovidat leat for me
19:17.18QwellDovid: Ask the bartender for 2 drinks at a time.  It makes the parties funner.
19:17.33QwellAlso, find the nearest Snom sales person after the party, and...
19:17.35Qwellsmirks
19:20.28Simon--hrm.. seeing an issue with debian wheezy's asterisk 1.8.13.1-based package.. all iax2 peers fail qualify over a minute or so, and won't come back until asterisk is restarted. ping (once per second) is fine the whole time, and the qualify in the opposite direction from remote boxes is still fine. sound like an old bug at all?
19:21.00Simon--I set up some stuff to capture iax2 netstats and threads..
19:21.50Qwell~upgrade asterisk
19:21.51infobotBefore requesting assistance, you should be running the latest version of a supported release branch.  See the channel topic for the latest versions available in currently supported branches.
19:23.44Simon--lols
19:25.14QwellSimon--: It was released 22 months ago.  It's not worth spending our time trying to figure out if it's already been fixed.
19:25.48Simon--sure. I was just wondering if anybody remembers such a bug
19:26.12Simon--there are some deadlock-related commits. I will track it down and see if they are related
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19:28.12Qwellhe has returned!
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19:29.02fileyeah, apparently IPv6 routing is a bit ... wonky
19:29.36Simon--is there an officialish svn->git mirror?
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19:37.15QwellSimon--: tzafrir runs(ran?) one on github I think.  I can't google right now, but it shouldn't be too hard to find.
19:37.31Qwellnot official, but it's probably the best you'll find
19:38.14tzafrirhttps://gitorious.org/asterisk-tools/asterisk
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19:56.46bananapieIs there a dial command that would cause newly bridged channels to mute for 5 seconds ( first five seconds ) before connecting the audio ? Transmitting silence would be ok.
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19:58.04newtonrbananapie, All you have to do is look through the dial help text, though I don't think there is.
19:58.15newtonrmaybe!
19:58.27bananapieI did look through dial help, I found nothing. A macro might work though.
19:58.36Qwellmacro/gosub should work fine
19:58.37bananapiewait
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20:10.50DovidSnom salesman.... Do i not want to ask?
20:14.28QwellDovid: You really don't.
20:14.38Dovidwill leave it at that
20:14.53Qwell(there were fighting words)
20:15.48[TK]D-Fendercomes from a place where ALL words are "fighting words"
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20:22.50fullstopapparently the obihai ATA's now support google voice (again)
20:23.07Qwellfullstop: officially
20:23.19fullstopyes
20:24.03fullstopdoes this mean anything, at all, for asterisk?
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20:26.45zerohalolol - I hope snom salesman doesn't come back
20:27.01Qwellzerohalo: Were you there?!
20:27.10zerohaloI was!
20:27.23zerohaloSaw it all go down.
20:28.06marceloamorimguys, there is any place too see if ccss(request) is requested?
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20:42.43marceloamorimlike /var/spool/asterisk/outgoing, maybe /var/spool/asterisk/ccss ;)
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21:14.47darkdrgn2k3what does [2014-09-11 16:13:21] WARNING[3083][C-0000000e]: channel.c:4860 ast_prod: Prodding channel 'SIP/8999-00000011' failed
21:14.47darkdrgn2k3<PROTECTED>
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