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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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10:59.49 | kangtastic | hi, 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? |
11:01.24 | file | no. |
11:10.14 | kangtastic | thanks. figured i'd ask just in case the docs were wrong |
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13:34.18 | igcewieling | What might prevent * from working during a voicemail greeting? |
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13:53.52 | Samot | Working how? |
13:53.58 | Samot | What stops working? |
13:55.03 | igcewieling | the answer is "if the exit context is empty then * out of voicemail won't work" |
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13:58.31 | [TK]D-Fender | igcewieling, you need an "a" in that context, not just being "not empty" |
14:00.35 | igcewieling | [TK]D-Fender: I'm using FreePBX so there is already an "a" extension in the macro-vm context. Good point though. |
14:00.37 | Samot | Ahhhh |
14:00.44 | Samot | "*" as in the actual use |
14:00.51 | Samot | Not "*" as in short of Asterisk.. |
14:01.54 | igcewieling | People who use * to mean "the asterisk pbx" are idiots. 8-| |
14:02.35 | Samot | So you don't want them to hit * or 0 during the greeting process? |
14:04.31 | igcewieling | They want to be able to press * during voicemail greeting to go into voicemail main. Thankfully I've fixed the issue. |
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14:31.25 | sibiria | in SDP how can i accurately pick out the RTP server used? |
14:31.32 | sibiria | is it always sent as root originator? |
14:32.33 | file | the c= line is for connection information, and the port is in each m= line |
14:32.42 | file | however each media stream may have its own c= line |
14:33.16 | sibiria | isn't the connection information optional in SDP? |
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14:33.48 | file | no. |
14:34.07 | file | https://tools.ietf.org/html/rfc4566#page-14 |
14:34.18 | Samot | Well |
14:34.21 | Samot | Yes and no. |
14:34.32 | Samot | It's not required in the Media if its in the Session |
14:34.34 | Samot | And vice versa. |
14:34.38 | sibiria | ah |
14:34.41 | Samot | There are two sections in which c= exists |
14:34.45 | Samot | It has to be in one of them. |
14:35.19 | Samot | Just keep in mind though, it might not be the actual RTP server. |
14:35.59 | Samot | Asterisk handles all my RTP but Kamailio runs RTPProxy so to the users all the media is coming from Kamailio |
14:36.44 | Samot | Which 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 |
14:39.22 | Samot | So just a little FYI to keep in mind. |
14:40.59 | Samot | On 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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17:47.35 | igcewieling | Does anyone happen to know if Adtran licensed their phones from Polycom? |
18:09.54 | jameswf | those types of things aren't typically "public" |
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18:20.15 | jameswf | That said 2+2 sometimes = 4 |
18:51.19 | igcewieling | jameswf: I'll know when I try to provision them with our internal provisioning gui which only works with Polycom phones.8-| |
18:52.43 | igcewieling | aha, I just received a picture of the boxes, they have Polycom MACs, even if they say Adtran on the label. |
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19:17.43 | jameswf | Yealink makes Verizon badged phones. The trick is do they have custom firmware that locks things |
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19:34.51 | jkroon | asterisk 13.22.0 - does anyone have any ideas why res_odbc takes 5 seconds (exactly) to reload? |
19:37.14 | file | attach gdb during that time and see where it is blocked? |
19:38.35 | jkroon | now why did I not think of that ... thanks file |
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19:49.49 | jkroon | so it's sporadic. really don't like those. |
19:51.03 | jkroon | @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. |
20:06.25 | jkroon | file, it seems to be slow when there are active connections to the database, and fast otherwise. |
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20:30.10 | P-NuT | Has 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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22:00.40 | Kobaz | meep |
22:00.47 | Kobaz | anyone familiar with sangoma vega pri gateways |
22:01.21 | Kobaz | getting a BYE Reason: Q.850 ;cause=102 |
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22:07.00 | shanth_ | what is the asterisk command to show what context a number will go to? i have used it before but didn't save it :( |
22:07.54 | shanth_ | 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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23:52.58 | sicelo | <PROTECTED> |
23:53.24 | sicelo | here is a paste from sip show peer: http://paste.debian.net/1043531/ |
23:54.02 | sicelo | the first client is the one giving me a problem. however, the second one, even using the credentials of the 'first' works just fine |
23:54.18 | sicelo | any ideas on what i am doing wrong? |