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03:07.51 | amitsinha280 | hi |
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03:09.21 | amitsinha280 | can anyone help me to complete configure asterisk for webrtc. i follow the docs but services not starting on 8089 port it got error |
03:09.52 | amitsinha280 | [May 6 08:17:38] WARNING[9671] cel_tds.c: cel_tds module had config problems; declining load[May 6 08:17:38] NOTICE[9671] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.[May 6 08:17:40] WARNING[9671] loader.c: Some non-required modules failed to load.[May 6 08:17:40] ERROR[9671] loader.c: cel_radius declined |
03:09.53 | amitsinha280 | to load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_tds declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_pgsql declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cel_sqlite3_custom declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_radius declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cel_tds declined to |
03:09.53 | amitsinha280 | load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_sqlite3_custom declined to load. |
03:13.37 | amitsinha280 | i am using asterisk v 17.4 |
03:16.16 | amitsinha280 | ?? |
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03:21.54 | amitsinha280 | please share any information regarding my issue |
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03:35.43 | amitsinha280 | drmessano yesterday you told to check the debug log for error. I checked and found some error but i can't find the solution to solve it. please help me |
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03:44.11 | drmessano | Pastebin the errors |
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04:06.50 | amitsinha280 | drmessano when i start server using /etc/init.d/asterisk start command in log file /var/log/asterisk messages i found error ... [May 6 08:17:38] WARNING[9671] cel_tds.c: cel_tds module had config problems; declining load[May 6 08:17:38] NOTICE[9671] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.[May 6 |
04:06.50 | amitsinha280 | 08:17:40] WARNING[9671] loader.c: Some non-required modules failed to load.[May 6 08:17:40] ERROR[9671] loader.c: cel_radius declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_tds declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_pgsql declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cel_sqlite3_custom declined to |
04:06.50 | amitsinha280 | load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_radius declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cel_tds declined to load.[May 6 08:17:40] ERROR[9671] loader.c: cdr_sqlite3_custom declined to load. |
04:08.54 | drmessano | amitsinha280: |
04:09.05 | drmessano | 1. I said pastebin, not "dump them in here" |
04:09.23 | drmessano | 2. We are looking for errors related to your issue, not CDR errors |
04:11.19 | drmessano | Also, pastebin your configuration |
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04:29.32 | drmessano | You're kidding, right |
04:30.04 | drmessano | RIP amishitna |
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09:21.01 | velix | When sttting Answer() |
09:21.01 | velix | <PROTECTED> |
09:25.40 | sibiria | i could be wrong but i don't think wait() sends silence |
09:25.49 | sibiria | also, you can tell answer() to wait after bridging the call |
09:26.03 | sibiria | e.g. Answer(1500) (milliseconds) |
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09:51.39 | velix | sibiria: I want to wait a second after answering. |
09:52.01 | velix | sibiria: I thought, Answer(1500) does wait until answering, oops. |
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10:20.46 | nibbier | hi. I operate a small asterisk connected to deutsche telekom for outside connections. we had problems that specific calls didn't work. calling in from many mobiles did not hear the asterisk users. other direction worked. network sniffing showed that if * sent out g711 streams it was not forwarded to the external phone by telekom. so i disabled alaw codec in sip.conf. this fixed this one problem, but now we could not talk to certain cities |
10:21.00 | nibbier | so I assume certain gateways of telekom need us to speak alaw, other need g722 - should this not be negotiated in SIP? |
10:21.35 | file | it is negotiated in SIP, specifically SDP |
10:22.10 | file | "here's the codecs I want to use" "well, here's what I've accepted" |
10:29.15 | sibiria | nibbier: have you considered offering alaw, ulaw and g722? just to cover all three common options |
10:29.35 | nibbier | indeed I see in the SDP for an incoming call "Media Format: ITU-T G.711 PCMA" (Wireshark interpretation). And if we then send G.711 it does not make it to the endpoint. In the "200 OK" packet I can't find any agreement, but more of a list of codecs my end would support. It seems asterisk chooses whatever he likes from the received list? |
10:30.03 | sibiria | g722 is becoming more common with european operators, and now and then u-law is seen within europe as well |
10:30.39 | nibbier | sibiria: the problem as it seems now is that telekom is offering something it then does not accept |
10:30.41 | file | Asterisk will allow what is common between the configuration and what was offered |
10:31.10 | file | would like to go one month without hearing of deutsche telekom |
10:31.12 | nibbier | file, ah okay.. makes sense |
10:31.56 | nibbier | is there importance in the order of codecs? can I tell asterisk what I would prefer him to send out? |
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10:32.15 | file | preference order is the order in the allow line |
10:32.35 | nibbier | great, then I'll go g722, alaw, ulaw.... let's try. |
10:39.45 | nibbier | seems to have fixed it. still shame on telekom to advertise stuff they won't support then... |
10:40.02 | nibbier | thanks sibiria, file |
10:40.22 | file | when it comes to VoIP they seem to do their own kind of thing... |
10:41.03 | nibbier | and no support at all.... all try & error with them. but I learned more about sip again, so well... |
10:41.21 | nibbier | and about wireshark, once again awesome :) |
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11:16.16 | Ionic | The bridge members were removed from ast_channel_tech, does that mean that bridging will have to be done exclusively through the Bridging API as introduced with Asterisk 12? |
11:16.42 | file | yes. |
11:21.15 | Ionic | The documentation seems to declare this as WIP, so that kept me wondering |
11:21.27 | file | which documentation? |
11:22.09 | Ionic | uhm, this one: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+API |
11:22.28 | file | ah that's a project page |
11:22.50 | Ionic | oh |
11:22.54 | file | it's a place where thoughts/initial stuff/etc went while kicking it off |
11:23.25 | Ionic | is there a proper place, or just the doxygen stuff and looking through the code? |
11:25.21 | file | we generally don't document the APIs and such on the wiki because they then go out of sync with the actual implementation, so it's just the source code |
11:26.31 | Ionic | okay, thanks. :) |
11:28.47 | Ionic | also... more of an informal question... is POTS and ISDN usage mostly dead these days? consumer hardware using the cologne chip is only really supported through mISDN and LCR with Asterisk integration through chan_lcr, although its code isn't updated very often. I've tried porting it to Asterisk 16 and it did look reasonably well until noticing that it used the old masquerade bridge feature which will |
11:28.53 | Ionic | likely need a half rewrite. |
12:14.24 | nibbier | telekom advertises Media Descriptios "RTP/AVP 96-103", which wireshark tells me is "DynamicRTP-Type-[96-103]" besides g711/g722 on some calls. what is this? |
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12:17.51 | file | it |
12:17.57 | file | erm, it's dynamic, would need full SDP to see |
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13:00.28 | nibbier | file: ah, okay. |
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13:11.04 | volga629_ | Hello Everyone, what var function I can use to capture on inbound call from header user name is it ${EXTEN} ? |
13:12.35 | sibiria | SIP_HEADER() i think it is, if you're using chan_sip |
13:12.48 | qermit | anoether question. does asterisk supports X-Forward-For on WSS and WS transports? |
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13:13.52 | sibiria | and probably PJSIP_HEADER() if using pjsip |
13:14.43 | volga629_ | chan sip |
13:15.16 | volga629_ | I am trying remove prefix 32 from specific inbound from numbers |
13:16.04 | file | qermit: no |
13:16.24 | sibiria | if you want to retrieve the number calling in then CALLERID(num) is a better choice |
13:16.49 | volga629_ | ok thanks let me try put some example |
13:18.27 | qermit | file: :/ |
13:22.03 | volga629_ | https://paste.centos.org/view/9ca43755 |
13:22.44 | volga629_ | that should take out 32 |
13:22.54 | volga629_ | for ACT_DID |
13:25.16 | volga629_ | but how actually change ${CALLERID(num)} if ACT_DID match regex ^(455|499) |
13:27.33 | sibiria | Set(CALLERID(num)=123) |
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13:35.00 | volga629_ | so I need first remote 32 set variable for actual number then set another variable and print first 3 digits then set if statement |
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14:07.58 | volga629_ | some thing like this |
14:08.00 | volga629_ | https://paste.centos.org/view/564cf032 |
14:08.44 | volga629_ | what you think |
14:12.43 | sibiria | the priorities are all wrong |
14:13.03 | sibiria | i presume you want to distort/anonymize the caller id to present just the area code? |
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14:18.12 | volga629_ | you mean priorities ,1 |
14:18.23 | volga629_ | it start from 0 ? |
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14:20.35 | volga629_ | or I need use 1, and then n, |
14:22.12 | igcewieling | you could look at extensions.conf.sample and see |
14:22.14 | sibiria | 1 and then n will suffice |
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14:32.24 | volga629_ | so it should be ok ? |
14:34.03 | sibiria | i would use GotoIf() |
14:35.01 | sibiria | (since you want to jump around in the dial plan to execute conditionally) |
14:35.21 | volga629_ | ok I will switch it |
14:55.12 | volga629_ | I will test soon waiting for maintenance window sibiria thank you for all help |
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14:58.06 | sibiria | you'd probably want to have a test setup for development purposes :) |
15:01.08 | volga629_ | :-) |
15:06.34 | velix | I'm originating a call by callscript. When the first called person declines the call, the callscript doesn't get removed, but retries after some seconds or minutes. Is there a way to stop this? |
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15:10.41 | Samot | Tell it to not retry. |
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15:12.26 | velix | Samot: oh |
15:12.28 | velix | ;) |
15:12.56 | velix | MaxRetries: 0 ? |
15:13.24 | velix | I've set this already :( |
15:22.10 | velix | I think, the student did something wrong. I cannot reproduce it. |
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17:50.04 | igcewieling | Verizon's secret logo: https://i1.wp.com/blog.flashrouters.com/wp-content/uploads/2018/08/nexus2cee_devil-728x352-728x3521.png |
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17:55.32 | Samot | Aren't they like your lifeblood? |
17:57.00 | igcewieling | Correct. I still hate them. |
17:59.26 | Samot | I mean there are solutions to that. |
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18:28.43 | cybrNaut | VZ are scumbags when it comes to consumer privacy, but at least they aren't as bad as AT&T. VZ also recently dropped ALEC membership, while AT&T remains a scumbag ALEC member |
18:29.47 | cybrNaut | ethics is always relative. T-mobile-Sprint is clearly the lesser of evils. If I had to choose 1 of the 3, it'd be Tmobile |
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18:32.12 | cybrNaut | VZ is also the only carrier who does not make it easy for just anyone to buy someone else's location tracking records |
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18:40.04 | igcewieling | I wasn't referring to Verizon Wireless, which is a different company than Verizon |
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19:00.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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19:03.05 | cusco | basically my question is: what is the name of the driver, when using res_pgsql to put in extconfig? |
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19:13.59 | seanbright | cusco: i assume you mean res_config_pgsql |
19:14.05 | seanbright | and then answer to your question is pgsql |
19:14.26 | cusco | yes it is and I did meant it |
19:14.40 | cusco | it connects at least the cli cmd realtime pgsql show status |
19:14.43 | cusco | shows it is connected |
19:14.48 | cusco | however sip peers are not read... |
19:15.06 | igcewieling | IIRC, they won't be read until the first device registers. |
19:15.08 | seanbright | chan_sip? |
19:15.14 | cusco | owww |
19:15.18 | cusco | yes old chan sip |
19:15.28 | seanbright | yeah i dunno then |
19:15.46 | igcewieling | if something registers does the registration work? |
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20:09.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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20:18.46 | igcewieling | cybrNaut: Verizon Wireless is a joint venture between Verizon and Vodaphone or one of those other European carriers, but more importantly Verizon (ILEC) and Verizon Wireless are regulated differently. |
20:19.08 | igcewieling | Verizon has universal service obligations. Verizon wireless does not. |
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