IRC log for #asterisk on 20210209

00:07.29catphishi'm still getting my head around kamailio again after no using it for a few years, but think i'm getting there :)
00:10.20catphishah lovely, pjsip can match arbitrary headers too, so i can mark incoming UA or PSTN invites however i like, marvellous
00:18.56SamotYou still needed dialplan
00:21.43SamotAnd you need individual endpoints
00:21.59SamotEspecially if you are doing voicemail
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00:37.25catphishthat's fine, i'm happy with how things will work in asterisk once everyone's authenticated
00:40.37catphishinitially the dialplan will be absurdly simple, voicemail retrieval, local numbers back to kamailio where the UA's are registered then voicemail on timeout, and everything else (valid) to PSTN
00:41.08catphishbut being able to do different contexts for proxied peers will likely make more complex stuff easier later
00:41.27catphishthanks again, it's all getting clearer
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01:33.42SamotYikes.
02:44.05igcewielingIf anyone runs Windows and feels like testing something for me.  http://help.nyigc.net/tmp/test.wav  does that play without audio artifacts?
02:45.15igcewielingIt is the default voice for google cloud TTS, in case any cares.
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03:37.11zguhmm, grandstream uses one SIP header for auto-answer and avaya uses a different one
03:37.50zguand you can't just send both, because the avaya phone will just completely ignore the invite if it has the header grandstream uses. so much for a simple page macro
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10:44.53wakko[Feb  9 11:37:38] ERROR[32098]: loader.c:2396 load_modules: Failed to resolve dependencies for res_pjproject
10:44.54wakko[Feb  9 11:37:38] ERROR[32098]: loader.c:2396 load_modules: res_pjproject declined to load.
10:45.03wakkois there a way to know which symbol/module is missing ?
10:45.28wakko(version 16.16)
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15:35.08KobazSome carrier aggregators are now announcing origination redundancy
15:35.10KobazI'm wondering how this is accomplished... something similar to a resporg?
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15:52.09SamotIn what way?
15:52.35SamotThere's a difference between me having my upstream send a call to multiple switches..
15:52.48SamotSo if Switch A goes down, it goes to Switch B
15:52.56mrkikoHello to all. I am trying to develope an open source GSM->SIP gateway using dongle modems; interacting with Modemmanager. So far, so good. Having some issues with RTP: I can read and write audio to serial with SOX with this command-line: play --buffer 320 -t raw -r 8k -e signed -b 16
15:53.31mrkikohow do I express this with RTP ? Or, what approach do you recommedn me ? I will need to read from serial and send out via rtp and read from rtp and write to serial.
15:54.04mrkikoWhat rtp library do you recommend? Trying to use ortp so far, but I am a little bit lost. Was thinking payload 0 pcmu/8000 was the right choice but it might nt
16:07.54mrkikoany help really welcome
16:09.52mrkikomight be a=rtpmap:3 gsm/8000 a viable option?
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16:23.22mrkikoDoes freeswitch support the rtpmap:3 gsm/8000 rtp payload format?
16:28.44mrkikooops, sorry, wrong channel. You know, I'm trying to research this information all around
16:28.56mrkikonot an expert, and would like my program to work with any pbx
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16:35.38igcewielingIf you are not an expert, then you should not be trying " trying to develope an open source GSM->SIP gateway using dongle modems"
16:36.04igcewielingThis will be HARD, like rock science hard.
16:42.49mrkikoigcewieling: well, so far things have been good
16:43.46mrkikoI'm not expert in the VIP field itself, I worked with other stuff but would like to try. In university I graduated in TLC Eng, I think I can understand :D
16:43.51mrkikoat least try
16:44.36mrkikoSo far I am using exosip to handle the SIP layer and SDP. Now was trying to understand the audio part of it.
16:45.01mrkikothat said, igcewieling, any ideas ??
16:46.25igcewielingBuy a Total Access 9xx and reverse engineer it.   lol!
16:47.13mrkikoigcewieling: :D ok
16:47.38mrkikowell, it has been little bit hard... but I would like to have it since the days I used asterisk_chan_dongle
16:47.42igcewielingI'm only partially joking.   You should be able to find them cheap and you can look at the actual SIP debug.
16:48.09igcewielingIt won't help with the GSM part, of course.
16:48.27mrkikoigcewieling: thanks for the hint! I was thinking aboutusing libsox to convert the payload to ulaw
16:48.45mrkikowell, the sip part is kinda sorted out in itself, I did read the RFC
16:48.50mrkikoto help me out
16:49.00igcewielingIf anyone runs Windows and feels like testing something for me.  http://help.nyigc.net/tmp/test.wav  does that play without audio artifacts?
16:49.02mrkikohttps://tools.ietf.org/html/rfc3666
16:49.24mrkikonot on windows, sorry
16:49.39igcewielingOn my single windows box I have accessable has audio issues anyway.
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17:41.24KobazSamot: in Origination way
17:41.45KobazSamot: DID 1231231234 is routed to Carrier1, say Verizon
17:41.49KobazVerizon has a massive outage
17:42.00KobazThere goes your number... you can move that DID to say, Level3
17:42.02Kobazand continue service
17:42.18KobazApparently that's a new thing now, "comming soon"
17:42.42KobazI know resporgs can move toll free's around for redundancy
17:43.10igcewielingWithout actual technical details, that is just marketing BS.
17:45.14Kobazjust curious how that could be accomplished
17:50.26igcewielingI imagine it involves playing with the LRN or something similar.
18:03.29Kobazsoooo moving onto more exciting things
18:03.56Kobazopus.... how do you tell pjsip to offer more than one rate for opus?  my SDP going out is opus/48000, yet i have others defined in codecs.conf
18:21.33Kobaztrying to get a yealink phone to set itself on 8000
18:23.28Kobaz(user on a cruddy connection)
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19:19.17seanbrightigcewieling: sounds fine to me
19:20.17igcewielingseanbright: thanks.  it is sln16 format with a riff header.   google cloud text-to-speech supports generating audio in that format.
19:24.10igcewielingAsterisk plays it as long as it has a .sln16 extension, but it gets an audio blip at the beginning.
19:26.50seanbrighthmm
19:27.02seanbrightwell, the blip is the riff header
19:27.44seanbrightigcewieling: have you tried a .wav16 extension with asterisk?
19:28.09igcewielingseanbright: wav16?  /me googles.
19:28.39seanbrightit's an asterisk thing
19:28.53seanbrighthttps://github.com/asterisk/asterisk/blob/master/formats/format_wav.c#L519
19:29.55igcewielingseanbright: added in Asterisk 10?
19:30.19seanbrightappears so
19:30.28seanbrighthmm, 1.8 i think
19:30.58seanbright02 Sep 2010
19:31.09seanbrightnot guaranteeing it will work, just something to test
19:31.31igcewielingwow, I really need to keep up to date on codecs.   I'll give it a try.
19:34.17filetechnically it's a file format, not a codec #pedantic
19:34.36seanbrighti was going to make the distinction too, but opted to let it go
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19:40.34seanbrightigcewieling: it's been around since 1.8.0 looks like
19:41.15seanbrightso "forever" for all intents and purposes
19:42.39igcewielingIn the past I stuck with ulaw/wav, g722 and sln16.   I never needed to find a high quality format which could play on windows and Asterisk without modification.
19:42.55igcewielingThis is frickin awesome!
19:43.13igcewielingBTW, using wav16 worked.
19:56.37seanbrightgood deal
19:59.18mrkikoregarding my codec-related question, any idea?
19:59.35mrkikoor am I better converting the audio in flight?
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20:12.08seanbrighti have no thoughts/ideas on your question, no
20:17.52igcewielingI avoid low level protocol stuff when I can.
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22:19.31mrkikosorry, I exited without saying goodbye. thank you guys for the help. Have a good night.
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