IRC log for #asterisk on 20170207

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00:13.21Kalaverahey guys, I have analized a RTP packet using wireshark and I see some red lines with the following status: Wrong sequence number
00:13.44Kalaveraany idea why would it be happening? I see the same seuqence number on the previous line
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00:26.42Kobazareee teee peeeeee
00:27.19KalaveraKobaz: whats teee peeeeee
00:27.47Kobazyou have jitter
00:27.51Kobazthat's basically what that means
00:27.56Kobazyou got a packet out of order
00:28.53Kalaverabut the recorded audio is not right , seems like one side speak over the other
00:29.06Kalaveraand there are a bunch of those errors
00:29.10Kalaveraon revert
00:30.08Kobazthat's going to be an artifact of delay
00:30.20Kobazyou have a laggy/lossy connection
00:36.20KalaveraKobaz: could be a route for an inexistent sip service?
00:36.26Kobazno
00:36.33Kobazyou have an existing media stream over a bad connection
00:36.45Kobazjitter will cause packets out of order
00:36.51Kobazdelay will cause people to talk over each other
00:37.02Kalaveraon the SIP Provider side?
00:37.11Kobazcould be anywhere
00:37.19Kobazuse mtr/ping/traceroute to find the problem spot
00:37.25Kalaveradoesnt matters if it is on Forward tab or revert tab ?
00:37.41Kalaveraalright thanks
00:37.49Kobazthis is a low level networking question, you need to determine where your connection problem is
00:38.01Kobazspeed test your network connection if it's via the internet
00:38.11Kobazand check on ping/mtr definitely
00:38.34Kobazyou can use iperf for server to server speed tests
00:38.41Kalaverayes it is via internet
00:38.54Kobazand use tcpdump/wireshark to check for retransmits and things like that
00:39.03Kobazcheck #networking for more help on troubleshooting lag/latency
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02:33.37igcewielingKobaz: does the problem if you swap out the user instead of the phone?
02:33.48igcewielingKobaz: does the problem happen if you swap out the user instead of the phone?
02:49.36igcewielingSwapping out the user has fixed at least two issues.   The first is the user was not doing the transfer correctly and the 2nd time the user was not seating the handset back in the cradle all the way.
02:50.21igcewielingdf -h
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04:05.18metanoviihi, i compile asterisk 13.13.1 with opus , and now i have this in log ERROR[5127] codec_opus_config.c: Opus: failed to register sorcery object type 'opus'  . Asterisk can not start.
04:05.48metanoviihow can i fix this?
04:09.42[TK]D-Fenderhttps://community.asterisk.org/t/asterisk-13-opus-codec/68179/13
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11:15.04WIMPy*Time
11:15.08WIMPyups
11:15.18WIMPyAny Fedora users here? https://plus.google.com/u/0/+ChristianMGrube/posts/MNB74LGWBvA
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12:52.41TandyUKanyone aware of a good tutorial for setting up homer with captagent 6 for logging rtcp-xr data only (for now)
12:53.02TandyUKgoogling just keep comign up with the same few pages, which arent really explaining anything!
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15:25.53ctjctjHello.  I'm trying to find the configuration where I set the voice to use.  Right now my asterisk is using sounds/en_US_f_Allison/  For certain contexts I need to set it to use en_US_f_Donna (our voice talent(?))
15:26.38[TK]D-Fender"core show function CHANNEL" <- set the language
15:26.49[TK]D-Fenderor move the files around
15:30.37tzafrir_"move files around" doesn't help when you need both sets to be used at the same time
15:31.16[TK]D-FenderDepends on overlap.
15:31.46tzafrir_ctjctj, BTW: is there no sounds/en that points to en_US_f_Allison ?
15:32.01[TK]D-FenderIf you intend to use specific files for each language at all times, then you can use "en_US_f_Donna" as the set language, and all the others in the default fallback en folder
15:32.50tzafrir_"en" happens to be the fallback. This happens to work for English, but not for any other language.
15:33.11tzafrir_But only works if the "en" symlink points to that directory.
15:33.49ctjctjtzafrir_, [TK]D-Fender thank you.
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15:34.15[TK]D-Fendertzafrir_, Or... you move files around ;)
15:35.07tzafrir_Anyway, a voice talent is better than a voice telnet.
15:36.14ctjctjOk.  It looks to me like I'll handle it differently.  Not a problem.
15:36.33ctjctjI was hoping for a "already there" answer.  And yes, "en" points to en_US_f_Allison.
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16:41.40wonderworldhi, where would i find documentation on http.conf with tls? i am unsure on how to set it up with SSL in a working manner. thanks a lot.
16:46.52[TK]D-Fenderin the sample config
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16:56.48avbwonderworld: its pretty straightforward, just add tls certs/keys and define a port
16:57.43wonderworld[TK]D-Fender: thanks, i found it, but i can't make asterisk listen on the SSL-port (standard http works fine). i guess something is wrong with my certs. is there a way to see log output on the mini httpservers startup?
16:58.53wonderworldi am trying to use letsencrypt-certs. they should work, shouldn't they?
17:00.12avbwonderworld: why dont you show your config first? :)
17:00.21wonderworldavb: yes, just a second
17:00.49avbas well, as a guess, are the persmissions on cert files are ok?
17:03.18wonderworldhttp://pastebin.ca/3764979
17:05.50avbhttp show status
17:05.55avbwhat does it shows
17:07.26avbtlsprivatekey=/root/.getssl/certs/asterisk_key.pem
17:07.38avbim pretty sure you got a permissions issue
17:07.50avbif your asterisk server is runned under asterisk user
17:08.20wonderworldhttp://pastebin.ca/3764981
17:08.23wonderworldok, let me check
17:09.01wonderworldsame thing when started as root
17:09.43wonderworlddocs say it needs .pem extension, so i created the softlinks for asterisk
17:09.55wonderworldmaybe i linked to the wrong files
17:10.23wonderworldi have CA_CERT DOMAIN_CERT DOMAIN_KEY and DOMAIN_PEM
17:10.42wonderworldasterisk_cert.pem links to CA_CERT
17:10.53wonderworldasterisk_key.pem links to DOMAIN_KEY
17:12.01wonderworldi tried to find any error message that would tell me why it fails to spawn the https part of the server but i was not successful yet
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17:25.55wonderworldok i fixed it.
17:26.13wonderworldyour asumption was nearly correct
17:26.22wonderworldhad a typo in the path to the cert
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17:26.28wonderworld*bangs head against wall
17:26.32wonderworldtnx a lot
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17:36.37avbnp
17:36.53avbstupid things are happening with everybody :)
17:37.02avbhates typos
17:37.19avbtrying to do things fast and in the end losing hours
17:42.33wonderworldso true
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18:06.49igcewielingI'm getting REALLY tired of Percona DB
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18:35.13avbigcewieling: whats wrong with it?
18:36.29igcewielingavb:  it causes every ODBC program I've tried to segfault.   apparently this has been happening on CentOS 7 for the last three releases of the Percona server.
18:37.58igcewielingI can work around it by getting the older RPM, extracting the older library, copy it to the right location and fix the symlinks.   Seems rather unprofessional for a show stopper bug to happen for so long.
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20:26.01sduthilhi, i've got a question about originate and callerid
20:26.01sduthilsuppose I have SIP phones A and B and I am implementing a Stasis application that does about the same thing as Dial (it does other things, but it's not relevant for this question)
20:26.01sduthilphone A calls Asterisk and gets in Stasis for my application
20:26.01sduthilmy application wants to call phone B and bridge A and B together
20:26.01sduthilso it originates phone B via ARI, with argument callerId=A
20:26.03sduthilphone B rings correctly, showing that A is the caller
20:26.05sduthilwhen phone B answers, it is directed to Stasi in my application again
20:26.07sduthilthen my application bridges A and B
20:26.09sduthilphone B still sees A as the correspondant, which is correct
20:26.11sduthilbut phone A sees A as the correspondant, because the CALLERID of B hasn't changed since the argument callerId= from the originate
20:26.14sduthilI've worked around that by adding some callerid logic in my application to restore the callerid of B when B answers
20:26.17sduthilMy question is: is there a better way to set the callerid of B after it answers? without adding logic to my stasis application? and without adding dialplan? is there some magic around CALLERID() that would allow that?
20:26.20sduthilmore generally, is there a way to control the callerid of the second part of an originate independently from the callerid of the first part?
20:29.19sduthilphone B does have a callerid configured in sip.conf, but it's overridden by the callerId= argument of the originate
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21:12.09Demon_VoIPAsterisk 13.14-rc2 (and early the same error). Voicemail via ODBC. res_mysql (mariadb). After VoicemailMain and press "1" asterisk hang OS and then go down. In /var/spool/asterisk/voicemail... INBOX i see file with 190GB size!!!  What's wrong?
21:12.40Demon_VoIPdescribe voicemail_messages: http://pastebin.com/n69XpEKf
21:14.17Demon_VoIPin "inbox" several messages. No more than some MB.
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21:24.09boomshakalakahowdy partners
21:25.13boomshakalakaI'm trying to install dahdi on my rasp pi running 4.4.38, but I keep getting an error that the current version doesn't support my kernel.
21:25.23boomshakalakaAnyone know which version of dahdi supports 4.4.38?
21:25.37boomshakalakaor better how I could find that information for myself
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21:59.59SamotWhy would you be installing DAHDI on a RasPi?
22:01.35SamotDAHDI is for when you have physical telephony hardware. PRI, FXS, FXO, etc.
22:01.53SamotNone of which, I believe, can be jammed into a RasPi.
22:02.02SamotWell and still function.
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