IRC log for #asterisk on 20180125

00:01.22josefighi, i know this is not elastix application, i need to setup a lower time between call and call on the call center application, but i don't know where to find it, can you give me a hand please? I'm using elastix
00:01.37josefig<PROTECTED>
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01:18.46*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
01:19.43n\ackOK, so restarting asterisk cleared the phantom pheature [sic]
01:21.40n\ackWTF?  Shouldn't dialplan reload and module reload features be the entirety??
01:22.18n\ackNa Ja.
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06:04.36BeeBuuhi,all. is webrtc make asterisk's performance low?
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06:20.34drmessanono
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08:32.15Aur313hi everyone
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08:35.41Aur313I'm using port 5070 for SIP, and I noticed that the "From" uri does not contain the port. this disturb some of my equipements... does anyone has any idea? (I tried from_domain but it adds some "[" "]" in around the address... and it doesn't seems good to me...))
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13:46.18jkroonsigh .. ok, so tcptlsserver.c at some point stops accept()ing new connections.
13:48.32jkroonwell, the main/tcptls.c code.
14:18.53mneth1Anyone around that could help with a stereo recording sync/alignment issue?
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14:20.57jkroonmneth1, g729?
14:21.12mneth1jkroon: g711u.
14:21.22mneth1it's sync in the stereo recording
14:21.41jkroonhmm, that doesn't have VAD afaik, which is what normally causes it in that case.
14:21.51jkroonmore specifically the CNG ...
14:22.23mneth1Are you familiar at all with this bug from the tracker? https://issues.asterisk.org/jira/browse/ASTERISK-26875?jql=text%20~%20%22mixmonitor%20sync%22
14:22.53mneth1well more specifically this: https://gerrit.asterisk.org/#/c/5209/
14:23.15jkrooninteresting.  never had that specific problem.  Used Monitor() mostly.
14:25.43Samotmenth1: What is the actual issue?
14:25.53Samotmneth1: ^^
14:27.11mneth1Samot: We're still having MixMonitor stereo recordings end up out of sync even after upgrading to 13.18.3, and it looks like that fix has been in since 13.16.
14:27.48SamotOK
14:27.56SamotSo the exact same thing?
14:28.06Samot183 with SDP but no RTP?
14:29.21mneth1I need to dig into more pcaps to confirm that, it does still seem linked to packet loss like in this: https://gerrit.asterisk.org/#/c/5209/
14:30.31SamotWEll..
14:31.14SamotI'm not saying you don't have problems. But I've seen plenty of problems and latch on to an old bug ticket because "it's like their issue".
14:31.37SamotSo if you are pointing to a specific bug issue, then that means you are having the exact same issues reported.
14:32.09SamotIf you are out of sync in other ways then focusing on the "I thought this was fixed already" line of thought is bad.
14:32.32mneth1Yeah for sure. I can't say for sure it's the exact same issue. Just researching where to look or what to test and I haven't been able to come up with much yet.
14:32.53SamotYou're also using PJSIP?
14:34.35mneth1Still using chan_sip right now. I do plan to try to migrate to pj_sip at some point though.
14:34.49SamotOk
14:34.50SamotSo
14:35.02SamotIt is in no why the same issue
14:36.02SamotIts hars to get help when you point at a known issue for people to look at then turns out youre not even in that type of setup
14:36.32SamotPlease provide actual details of your issue
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14:39.22mneth1Hm. We're using 13.18.3, chan_sip, and MixMonitor's t and r options to create separate recording files for the two stereo channels but in some instances the two recordings are ending up as different lengths. This makes it so when the two recording are merged into one, the conversations get out of sync/alignment. I've tried with the MixMonitor b option as well and the issue was still present.
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16:20.35wim_Hello, is it normal that when using pjsip with path support that when using contact sip:s@x.x.x.x it doesn't seem to work?
16:22.01wim_The Asterisk was receiving a REGISTER with Supported: path and a Path header, but when asterisk tried to send a qualify, it would ignore the path and not add a route header
16:23.09wim_when i changed the registration on the other end to have a contact with sip:sip_user@x.x.x.x it now work. It does't seem to like the contact s@x.x.x.x
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17:32.56imcdonaWhat determines if Asterisk is going to install default conf files in /etc/asterisk/ ? Is it merely the presence of any files /etc/asterisk/ ? There's no make target that says "don't put any config files in /etc/asterisk "
18:07.00rmudgettimcdona: Doing "make samples" puts the sample config files in /etc/asterisk
18:07.22rmudgettOtherwise you have to put files there.
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22:51.52Kobazis there a way to run a single line of dialplan
22:52.15Kobazsay you have a macro/context/etc, and you want to jump right into a specific label, run that code, and then jump back
22:52.27Kobazwithout modifying the original macro/context
22:52.49KobazGoSubSingleLine(foobar,baz,labelname)
22:53.02Kobazi guess i could write one
22:53.50Kobazmaybe just a general introspection would be useful
22:54.25Kobazlike, Set(dialplan=${READ_DIALPLAN(exten.context,prio)})
22:54.41[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Gosub
22:54.48[TK]D-FenderOr ... you could just USE Gosub
22:55.09Kobazright, i know how to use gosub, i'm talking about a specific-case
22:55.30[TK]D-FenderKobaz> say you have a macro/context/etc, and you want to jump right into a specific label, run that code, and then jump back <- it already DOES that
22:55.37Kobazno it does not
22:55.44Kobazyou need to explicitly Return();
22:55.48[TK]D-FenderSingle line?  No.  Jump to specific point, sure
22:55.55Kobazor implicitly run out of dialplan in that context
22:56.18Kobazrun a single line, and jump back, without any existing modifications
22:56.29[TK]D-FenderThere is no "single line".
22:56.32Kobazsure there is
22:56.34[TK]D-FenderTHere is "go", and you go.
22:56.47Kobaz<PROTECTED>
22:56.50Kobazthere's your single line
22:56.53[TK]D-Fenderno...
22:56.54Kobazor whatever
22:57.05[TK]D-Fenderthere is no way to jump BACK if thtere ar other priorities after it
22:57.10Kobazright, didn't think so
22:57.21Kobazso i like my READ_DIALPLAN idea the best
22:57.25Kobazit's the most flexible
22:57.34Kobazno jumping needed... and you can read as many lines as you like
22:58.03Kobazbasically just a quick query to confirm if that exists or not, if it was something i may have overlooked
22:58.13[TK]D-FenderNo.
22:58.31[TK]D-FenderWhat actual usage scenarion do you see as making this worth creating?
22:58.41Kobazworking with vendor-specific asterisk platforms
22:58.50Kobazwriting custom code for said platforms
22:59.06[TK]D-FenderSo only to hack other peoples "moving target" code....
22:59.22Kobazif need be
22:59.23[TK]D-Fenderyeah, * isn't built with trying to get competing code to cooperate
22:59.41Kobazright, that's not the intention
22:59.46Kobazbut there's no harm in giving the tools
23:00.00Kobaza READ_DIALPLAN type thing would work well for my own platform
23:00.16Kobazwhere sometimes i generate dialplan to be used by other bits of dialplan, and i only need some snippits
23:00.29[TK]D-FenderYou COULD hack this up in external scripting
23:00.45Kobazlike hunt groups, i have lists of hunt group members, and if i want member 1, i can gosub to member 1, which does a Set() Return();
23:00.47Kobazand then i have my value
23:01.16Kobazgrabbing dialplan would be a lot more direct
23:01.30Kobazyou could hack anything any which way
23:01.32Kobazdoesn't make it a good idea
23:02.06[TK]D-FenderYou're hacking someone else's GUI ... so yeah, swimming upstream.
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23:23.57kambeiI'm having a little difficulty with an Asterisk machine using a  Sangoma  A200 with a daughterboard, which should provide access to 8 POTS lines.  When I run the config, I'm seeing only two channels.  What am I doing wrong?  Does channell == POTS line?
23:24.20kambeiI'm far from adept when it comes to Asterisk, needless to say.
23:26.07kambeiI have my system.conf in /etc/dahdi which is showing fxsks 1 and 2, and those are reflected in the configs in /etc/asterisk ... I seem to have access to only two lines, or am I misunderstanding?
23:26.20[TK]D-Fendera defined channel is a single port in the case of a card like that
23:26.27[TK]D-FenderShow us your configs and the status dump
23:26.31[TK]D-Fender~pb
23:26.32infobotsomebody said pastebin was 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
23:26.33[TK]D-Fender^^^
23:27.45kambeihttps://pastebin.ca/3964211  <-  /etc/dahdi/system.conf
23:28.03kambei02:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card
23:28.17[TK]D-FenderYup, that's 2 channels
23:28.30[TK]D-FenderSo if you have more you expect to use... they're not configured
23:29.05kambei[TK]D-Fender: What am I doing wrong?  I was using the wanrouter config tools to generate this, and that's what I got
23:29.24kambei[TK]D-Fender: Why in particular is it detecting only two channels?
23:29.27[TK]D-FenderThat file only specifies 2 channels
23:29.33[TK]D-FenderGo change it
23:29.45kambeiBut that's what was generated.  Why did it generate two channels?  Why not one or three?
23:30.16[TK]D-FenderI don['t know what it think's it's detecting, but we shouldn't have to care.  You know what you bought right?  You know you have the right modules to support more channels on it?
23:30.32kambeiYes
23:30.45[TK]D-Fenderso just fix your config and specify the rest
23:31.15kambeiWhen I try to add more channels, I get this...
23:31.41kambeihttp://pastebin.ca/3964212
23:35.02[TK]D-Fenderwanpipe configs too.....

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