IRC log for #asterisk on 20200507

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03:07.51amitsinha280hi
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03:09.21amitsinha280can anyone help me to complete configure asterisk for webrtc. i follow the docs but services not starting on 8089 port it got error
03:09.52amitsinha280[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
03:09.53amitsinha280to 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
03:09.53amitsinha280load.[May  6 08:17:40] ERROR[9671] loader.c: cdr_sqlite3_custom declined to load.
03:13.37amitsinha280i am using asterisk v 17.4
03:16.16amitsinha280??
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03:21.54amitsinha280please share any information regarding my issue
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03:35.43amitsinha280drmessano 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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03:44.11drmessanoPastebin the errors
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04:06.50amitsinha280drmessano 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
04:06.50amitsinha28008:17:40] 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
04:06.50amitsinha280load.[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.
04:08.54drmessanoamitsinha280:
04:09.05drmessano1. I said pastebin, not "dump them in here"
04:09.23drmessano2. We are looking for errors related to your issue, not CDR errors
04:11.19drmessanoAlso, pastebin your configuration
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04:29.32drmessanoYou're kidding, right
04:30.04drmessanoRIP amishitna
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09:21.01velixWhen sttting Answer()
09:21.01velix<PROTECTED>
09:25.40sibiriai could be wrong but i don't think wait() sends silence
09:25.49sibiriaalso, you can tell answer() to wait after bridging the call
09:26.03sibiriae.g. Answer(1500) (milliseconds)
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09:51.39velixsibiria: I want to wait a second after answering.
09:52.01velixsibiria: I thought, Answer(1500) does wait until answering, oops.
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10:20.46nibbierhi. 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
10:21.00nibbierso I assume certain gateways of telekom need us to speak alaw, other need g722 - should this not be negotiated in SIP?
10:21.35fileit is negotiated in SIP, specifically SDP
10:22.10file"here's the codecs I want to use" "well, here's what I've accepted"
10:29.15sibirianibbier: have you considered offering alaw, ulaw and g722? just to cover all three common options
10:29.35nibbierindeed 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?
10:30.03sibiriag722 is becoming more common with european operators, and now and then u-law is seen within europe as well
10:30.39nibbiersibiria: the problem as it seems now is that telekom is offering something it then does not accept
10:30.41fileAsterisk will allow what is common between the configuration and what was offered
10:31.10filewould like to go one month without hearing of deutsche telekom
10:31.12nibbierfile, ah okay.. makes sense
10:31.56nibbieris there importance in the order of codecs? can I tell asterisk what I would prefer him to send out?
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10:32.15filepreference order is the order in the allow line
10:32.35nibbiergreat, then I'll go g722, alaw, ulaw.... let's try.
10:39.45nibbierseems to have fixed it. still shame on telekom to advertise stuff they won't support then...
10:40.02nibbierthanks sibiria, file
10:40.22filewhen it comes to VoIP they seem to do their own kind of thing...
10:41.03nibbierand no support at all.... all try & error with them. but I learned more about sip again, so well...
10:41.21nibbierand about wireshark, once again awesome :)
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11:16.16IonicThe 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?
11:16.42fileyes.
11:21.15IonicThe documentation seems to declare this as WIP, so that kept me wondering
11:21.27filewhich documentation?
11:22.09Ionicuhm, this one: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Bridging+API
11:22.28fileah that's a project page
11:22.50Ionicoh
11:22.54fileit's a place where thoughts/initial stuff/etc went while kicking it off
11:23.25Ionicis there a proper place, or just the doxygen stuff and looking through the code?
11:25.21filewe 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
11:26.31Ionicokay, thanks. :)
11:28.47Ionicalso... 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
11:28.53Ioniclikely need a half rewrite.
12:14.24nibbiertelekom 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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12:17.51fileit
12:17.57fileerm, it's dynamic, would need full SDP to see
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13:00.28nibbierfile: ah, okay.
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13:11.04volga629_Hello Everyone, what var function I can use to capture on inbound call from header user name is it ${EXTEN} ?
13:12.35sibiriaSIP_HEADER() i think it is, if you're using chan_sip
13:12.48qermitanoether question. does asterisk supports X-Forward-For on WSS and WS transports?
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13:13.52sibiriaand probably PJSIP_HEADER() if using pjsip
13:14.43volga629_chan sip
13:15.16volga629_I am trying remove prefix 32 from specific inbound from numbers
13:16.04fileqermit: no
13:16.24sibiriaif you want to retrieve the number calling in then CALLERID(num) is a better choice
13:16.49volga629_ok thanks let me try put some example
13:18.27qermitfile: :/
13:22.03volga629_https://paste.centos.org/view/9ca43755
13:22.44volga629_that should take out 32
13:22.54volga629_for ACT_DID
13:25.16volga629_but how actually change ${CALLERID(num)} if ACT_DID match regex ^(455|499)
13:27.33sibiriaSet(CALLERID(num)=123)
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13:35.00volga629_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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14:07.58volga629_some thing like this
14:08.00volga629_https://paste.centos.org/view/564cf032
14:08.44volga629_what you think
14:12.43sibiriathe priorities are all wrong
14:13.03sibiriai presume you want to distort/anonymize the caller id to present just the area code?
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14:18.12volga629_you mean priorities ,1
14:18.23volga629_it start from 0 ?
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14:20.35volga629_or I need use 1, and then n,
14:22.12igcewielingyou could look at extensions.conf.sample and see
14:22.14sibiria1 and then n will suffice
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14:32.24volga629_so it should be ok ?
14:34.03sibiriai would use GotoIf()
14:35.01sibiria(since you want to jump around in the dial plan to execute conditionally)
14:35.21volga629_ok I will switch it
14:55.12volga629_I will test soon waiting for maintenance window sibiria thank you for all help
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14:58.06sibiriayou'd probably want to have a test setup for development purposes :)
15:01.08volga629_:-)
15:06.34velixI'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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15:10.41SamotTell it to not retry.
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15:12.26velixSamot: oh
15:12.28velix;)
15:12.56velixMaxRetries: 0 ?
15:13.24velixI've set this already :(
15:22.10velixI think, the student did something wrong. I cannot reproduce it.
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17:50.04igcewielingVerizon's secret logo: https://i1.wp.com/blog.flashrouters.com/wp-content/uploads/2018/08/nexus2cee_devil-728x352-728x3521.png
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17:55.32SamotAren't they like your lifeblood?
17:57.00igcewielingCorrect.  I still hate them.
17:59.26SamotI mean there are solutions to that.
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18:28.43cybrNautVZ 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
18:29.47cybrNautethics 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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18:32.12cybrNautVZ 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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18:40.04igcewielingI 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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19:03.05cuscobasically my question is: what is the name of the driver, when using res_pgsql to put in extconfig?
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19:13.59seanbrightcusco: i assume you mean res_config_pgsql
19:14.05seanbrightand then answer to your question is pgsql
19:14.26cuscoyes it is and I did meant it
19:14.40cuscoit connects at least the cli cmd realtime pgsql show status
19:14.43cuscoshows it is connected
19:14.48cuscohowever sip peers are not read...
19:15.06igcewielingIIRC, they won't be read until the first device registers.
19:15.08seanbrightchan_sip?
19:15.14cuscoowww
19:15.18cuscoyes old chan sip
19:15.28seanbrightyeah i dunno then
19:15.46igcewielingif 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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20:18.46igcewielingcybrNaut: 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.
20:19.08igcewielingVerizon has universal service obligations.  Verizon wireless does not.
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21:47.12*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
22:35.27*** join/#asterisk matrix1233 (~matrix123@2a04:cec0:11e1:389f:1c63:5a13:88c2:29ab)
22:58.30*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
23:02.15*** join/#asterisk FolkLore (~The@190.103.179.211)
23:31.45*** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo)
23:45.08*** join/#asterisk Janos (~Janos@201.204.94.76)
23:49.38*** join/#asterisk ilius (~froy@slc.taskeasy.com)

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