IRC log for #asterisk on 20180920

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03:01.00*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.0 (2018/09/05) 16.0.0-rc2 (2018/09/12), Standard: 15.6.0 (2018/09/05); 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
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10:59.49kangtastichi, suppose i'm using PJSIP with an endpoint that has an auth. is it possible to allow unregistrations (REGISTER with Expires: 0) if they come from a trusted IP?
11:01.24fileno.
11:10.14kangtasticthanks. figured i'd ask just in case the docs were wrong
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13:34.18igcewielingWhat might prevent * from working during a voicemail greeting?
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13:53.52SamotWorking how?
13:53.58SamotWhat stops working?
13:55.03igcewielingthe answer is "if the exit context is empty then * out of voicemail won't work"
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13:58.31[TK]D-Fenderigcewieling, you need an "a" in that context, not just being "not empty"
14:00.35igcewieling[TK]D-Fender:  I'm using FreePBX so there is already an "a" extension in the macro-vm context.   Good point though.
14:00.37SamotAhhhh
14:00.44Samot"*" as in the actual use
14:00.51SamotNot "*" as in short of Asterisk..
14:01.54igcewielingPeople who use * to mean "the asterisk pbx" are idiots. 8-|
14:02.35SamotSo you don't want them to hit * or 0 during the greeting process?
14:04.31igcewielingThey want to be able to press * during voicemail greeting to go into voicemail main.    Thankfully I've fixed the issue.
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14:31.25sibiriain SDP how can i accurately pick out the RTP server used?
14:31.32sibiriais it always sent as root originator?
14:32.33filethe c= line is for connection information, and the port is in each m= line
14:32.42filehowever each media stream may have its own c= line
14:33.16sibiriaisn't the connection information optional in SDP?
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14:33.48fileno.
14:34.07filehttps://tools.ietf.org/html/rfc4566#page-14
14:34.18SamotWell
14:34.21SamotYes and no.
14:34.32SamotIt's not required in the Media if its in the Session
14:34.34SamotAnd vice versa.
14:34.38sibiriaah
14:34.41SamotThere are two sections in which c= exists
14:34.45SamotIt has to be in one of them.
14:35.19SamotJust keep in mind though, it might not be the actual RTP server.
14:35.59SamotAsterisk handles all my RTP but Kamailio runs RTPProxy so to the users all the media is coming from Kamailio
14:36.44SamotWhich is doing nothing more than just proxying it from the Asterisk boxes. So I don't have to give the end users multiple IPs RTP could come from because to them it all comes from the same IP
14:39.22SamotSo just a little FYI to keep in mind.
14:40.59SamotOn the flip side, it was the same way to Asterisk. Asterisk never actually sees the end users IP in the RTP because it's proxied through Kamailio.
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17:47.35igcewielingDoes anyone happen to know if Adtran licensed their phones from Polycom?
18:09.54jameswfthose types of things aren't typically "public"
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18:20.15jameswfThat said 2+2 sometimes = 4
18:51.19igcewielingjameswf: I'll know when I try to provision them with our internal provisioning gui which only works with Polycom phones.8-|
18:52.43igcewielingaha, I just received a picture of the boxes, they have Polycom MACs, even if they say Adtran on the label.
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19:17.43jameswfYealink makes Verizon badged phones. The trick is do they have custom firmware that locks things
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19:34.51jkroonasterisk 13.22.0 - does anyone have any ideas why res_odbc takes 5 seconds (exactly) to reload?
19:37.14fileattach gdb during that time and see where it is blocked?
19:38.35jkroonnow why did I not think of that ... thanks file
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19:49.49jkroonso it's sporadic.  really don't like those.
19:51.03jkroon@file, I still need to gather that information for you on the debug stuff on pjsip fresh start.  I could however (and i've not spent hours on this) figure out how to enable debugging in asterisk before modules start loading, I did spot the -d a little earlier whilst looking at other stuff, so am assuming that I can use -ddddddddd to the startup command.
20:06.25jkroonfile, it seems to be slow when there are active connections to the database, and fast otherwise.
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20:30.10P-NuTHas anyone configured asterisk with Sipgate?
21:21.19*** topic/#asterisk by rmudgett -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc2 (2018/09/12), Standard: 15.6.1 (2018/09/20); 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
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22:00.40Kobazmeep
22:00.47Kobazanyone familiar with sangoma vega pri gateways
22:01.21Kobazgetting a BYE Reason: Q.850 ;cause=102
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22:07.00shanth_what is the asterisk command to show what context a number will go to? i have used it before but didn't save it :(
22:07.54shanth_nvm i found it dialplan show
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22:31.12*** topic/#asterisk by rmudgett -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); 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
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23:52.58sicelo<PROTECTED>
23:53.24sicelohere is a paste from sip show peer: http://paste.debian.net/1043531/
23:54.02sicelothe first client is the one giving me a problem. however, the second one, even using the credentials of the 'first' works just fine
23:54.18siceloany ideas on what i am doing wrong?

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