00:07.29 | catphish | i'm still getting my head around kamailio again after no using it for a few years, but think i'm getting there :) |
00:10.20 | catphish | ah lovely, pjsip can match arbitrary headers too, so i can mark incoming UA or PSTN invites however i like, marvellous |
00:18.56 | Samot | You still needed dialplan |
00:21.43 | Samot | And you need individual endpoints |
00:21.59 | Samot | Especially if you are doing voicemail |
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00:37.25 | catphish | that's fine, i'm happy with how things will work in asterisk once everyone's authenticated |
00:40.37 | catphish | initially 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 |
00:41.08 | catphish | but being able to do different contexts for proxied peers will likely make more complex stuff easier later |
00:41.27 | catphish | thanks again, it's all getting clearer |
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01:33.42 | Samot | Yikes. |
02:44.05 | igcewieling | If anyone runs Windows and feels like testing something for me. http://help.nyigc.net/tmp/test.wav does that play without audio artifacts? |
02:45.15 | igcewieling | It is the default voice for google cloud TTS, in case any cares. |
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03:37.11 | zgu | hmm, grandstream uses one SIP header for auto-answer and avaya uses a different one |
03:37.50 | zgu | and 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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10:44.53 | wakko | [Feb 9 11:37:38] ERROR[32098]: loader.c:2396 load_modules: Failed to resolve dependencies for res_pjproject |
10:44.54 | wakko | [Feb 9 11:37:38] ERROR[32098]: loader.c:2396 load_modules: res_pjproject declined to load. |
10:45.03 | wakko | is there a way to know which symbol/module is missing ? |
10:45.28 | wakko | (version 16.16) |
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15:35.08 | Kobaz | Some carrier aggregators are now announcing origination redundancy |
15:35.10 | Kobaz | I'm wondering how this is accomplished... something similar to a resporg? |
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15:52.09 | Samot | In what way? |
15:52.35 | Samot | There's a difference between me having my upstream send a call to multiple switches.. |
15:52.48 | Samot | So if Switch A goes down, it goes to Switch B |
15:52.56 | mrkiko | Hello 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 |
15:53.31 | mrkiko | how 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. |
15:54.04 | mrkiko | What 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 |
16:07.54 | mrkiko | any help really welcome |
16:09.52 | mrkiko | might be a=rtpmap:3 gsm/8000 a viable option? |
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16:23.22 | mrkiko | Does freeswitch support the rtpmap:3 gsm/8000 rtp payload format? |
16:28.44 | mrkiko | oops, sorry, wrong channel. You know, I'm trying to research this information all around |
16:28.56 | mrkiko | not an expert, and would like my program to work with any pbx |
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16:35.38 | igcewieling | If you are not an expert, then you should not be trying " trying to develope an open source GSM->SIP gateway using dongle modems" |
16:36.04 | igcewieling | This will be HARD, like rock science hard. |
16:42.49 | mrkiko | igcewieling: well, so far things have been good |
16:43.46 | mrkiko | I'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 |
16:43.51 | mrkiko | at least try |
16:44.36 | mrkiko | So far I am using exosip to handle the SIP layer and SDP. Now was trying to understand the audio part of it. |
16:45.01 | mrkiko | that said, igcewieling, any ideas ?? |
16:46.25 | igcewieling | Buy a Total Access 9xx and reverse engineer it. lol! |
16:47.13 | mrkiko | igcewieling: :D ok |
16:47.38 | mrkiko | well, it has been little bit hard... but I would like to have it since the days I used asterisk_chan_dongle |
16:47.42 | igcewieling | I'm only partially joking. You should be able to find them cheap and you can look at the actual SIP debug. |
16:48.09 | igcewieling | It won't help with the GSM part, of course. |
16:48.27 | mrkiko | igcewieling: thanks for the hint! I was thinking aboutusing libsox to convert the payload to ulaw |
16:48.45 | mrkiko | well, the sip part is kinda sorted out in itself, I did read the RFC |
16:48.50 | mrkiko | to help me out |
16:49.00 | igcewieling | If anyone runs Windows and feels like testing something for me. http://help.nyigc.net/tmp/test.wav does that play without audio artifacts? |
16:49.02 | mrkiko | https://tools.ietf.org/html/rfc3666 |
16:49.24 | mrkiko | not on windows, sorry |
16:49.39 | igcewieling | On my single windows box I have accessable has audio issues anyway. |
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17:41.24 | Kobaz | Samot: in Origination way |
17:41.45 | Kobaz | Samot: DID 1231231234 is routed to Carrier1, say Verizon |
17:41.49 | Kobaz | Verizon has a massive outage |
17:42.00 | Kobaz | There goes your number... you can move that DID to say, Level3 |
17:42.02 | Kobaz | and continue service |
17:42.18 | Kobaz | Apparently that's a new thing now, "comming soon" |
17:42.42 | Kobaz | I know resporgs can move toll free's around for redundancy |
17:43.10 | igcewieling | Without actual technical details, that is just marketing BS. |
17:45.14 | Kobaz | just curious how that could be accomplished |
17:50.26 | igcewieling | I imagine it involves playing with the LRN or something similar. |
18:03.29 | Kobaz | soooo moving onto more exciting things |
18:03.56 | Kobaz | opus.... 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 |
18:21.33 | Kobaz | trying to get a yealink phone to set itself on 8000 |
18:23.28 | Kobaz | (user on a cruddy connection) |
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19:19.17 | seanbright | igcewieling: sounds fine to me |
19:20.17 | igcewieling | seanbright: thanks. it is sln16 format with a riff header. google cloud text-to-speech supports generating audio in that format. |
19:24.10 | igcewieling | Asterisk plays it as long as it has a .sln16 extension, but it gets an audio blip at the beginning. |
19:26.50 | seanbright | hmm |
19:27.02 | seanbright | well, the blip is the riff header |
19:27.44 | seanbright | igcewieling: have you tried a .wav16 extension with asterisk? |
19:28.09 | igcewieling | seanbright: wav16? /me googles. |
19:28.39 | seanbright | it's an asterisk thing |
19:28.53 | seanbright | https://github.com/asterisk/asterisk/blob/master/formats/format_wav.c#L519 |
19:29.55 | igcewieling | seanbright: added in Asterisk 10? |
19:30.19 | seanbright | appears so |
19:30.28 | seanbright | hmm, 1.8 i think |
19:30.58 | seanbright | 02 Sep 2010 |
19:31.09 | seanbright | not guaranteeing it will work, just something to test |
19:31.31 | igcewieling | wow, I really need to keep up to date on codecs. I'll give it a try. |
19:34.17 | file | technically it's a file format, not a codec #pedantic |
19:34.36 | seanbright | i was going to make the distinction too, but opted to let it go |
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19:40.34 | seanbright | igcewieling: it's been around since 1.8.0 looks like |
19:41.15 | seanbright | so "forever" for all intents and purposes |
19:42.39 | igcewieling | In 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. |
19:42.55 | igcewieling | This is frickin awesome! |
19:43.13 | igcewieling | BTW, using wav16 worked. |
19:56.37 | seanbright | good deal |
19:59.18 | mrkiko | regarding my codec-related question, any idea? |
19:59.35 | mrkiko | or am I better converting the audio in flight? |
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20:12.08 | seanbright | i have no thoughts/ideas on your question, no |
20:17.52 | igcewieling | I avoid low level protocol stuff when I can. |
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22:19.31 | mrkiko | sorry, I exited without saying goodbye. thank you guys for the help. Have a good night. |
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