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| 13:51.25 | jdean141 | Hi! I'm trying to write an asterisk module that'll use the dahdi channel (analouge interface card) to play pcm audio.\ |
| 13:51.57 | jdean141 | I'm having trouble finding any documentation on this, and not really sure where to start, I'd really appreciate a few pointers. Thanks! |
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| 15:51.32 | cervajs2 | hi. is there someone for discussing https://issues.asterisk.org/jira/browse/ASTERISK-24602 ? |
| 16:09.54 | cervajs2 | if 58 - AST_CAUSE_BEARERCAPABILITY_NOTAVAIL what can be problem? howto debug? |
| 16:10.32 | file | that will occur due to SDP negotiation failure |
| 16:10.44 | file | looking at both SDPs and figuring out what is wrong would be how to debug it |
| 16:11.05 | file | otherwise you have to follow the code |
| 16:17.31 | cervajs2 | thanks. we will try add detection what is bad in sdp. |
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| 16:24.25 | jsmith | jdean141: I'm not sure I understand what you're trying to do |
| 16:24.58 | jsmith | jdean141: You want it to play pcm audio during a call? What makes you think you need to write a new module? |
| 16:25.05 | jsmith | seems to be missing a big piece of the puzzle |
| 16:39.03 | putnopvut | I'm seeing something annoying/weird. |
| 16:39.22 | putnopvut | I have MALLOC_DEBUG enabled so I can look at memory allocations. |
| 16:39.39 | putnopvut | However, I'm also noticing that a lot of allocations are attributed to astobj2.c's __ao2_alloc() function |
| 16:39.57 | putnopvut | However, that's supposed to not get called if MALLOC_DEBUG is enabled, so I'm curious how that's showing up in the list. |
| 16:41.40 | file | hrm |
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| 17:21.09 | cervajs2 | file: we identified the problem is in /* Transport type is different in the remote answer. */ |
| 17:21.09 | cervajs2 | #define PJMEDIA_SDPNEG_EINVANSTP (PJMEDIA_ERRNO_START+46) /* 220046 */ |
| 17:21.59 | cervajs2 | it looks this is still problem with bad TLS detection in PJSIP. we are investigating further |
| 17:22.17 | file | SDP has nothing to do with the SIP transport |
| 17:23.03 | file | it's carried in SIP messages, but it doesn't care whether those are conveyed over UDP/TCP/TLS |
| 17:23.21 | file | chances are that the PJSIP configuration isn't right, causing it to not generate the expected SDP to negotiate |
| 17:24.49 | cervajs2 | can you check the pjsip.conf if i paste it to pastebin? |
| 17:25.13 | cervajs2 | its short |
| 17:26.05 | file | I'm about to walk home but https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 has a config, otherwise you have to look at the SDPs to understand what is going on |
| 17:27.40 | cervajs2 | tutorial is only for ws. ok we will try go deeper |
| 17:30.50 | cervajs2 | pjsip.conf http://pastebin.com/8Es4sDNp |
| 17:39.22 | jameswf | Astricon hotel have a shuttle? |
| 17:43.05 | file | cervajs2, nothing stands out |
| 17:46.12 | cervajs2 | SIP/2.0 200 OK Via: SIP/2.0/WSS 1.1.1.1:54635;rport=54635;branch=z9hG4bKPj655882f4-4910-43ce-928f-f55f94411a48;alias |
| 17:46.12 | cervajs2 | From: "webrtc_tomec"<sip:webrtc_tomec@2.2.2.2>;tag=12ab3743-1fea-4384-956a-878dd8ff9332 |
| 17:46.12 | cervajs2 | To: <sips:webrtc_cervenka@1.1.1.1;rtcweb-breaker=no>;tag=HThTgwNU3yDluwBCu8zA |
| 17:46.12 | cervajs2 | Contact: <sips:webrtc_cervenka@df7jal23ls0d.invalid;transport=wss> |
| 17:46.42 | cervajs2 | this is the problematic answer. is it ok that From: "webrtc_tomec"<sip:webrtc_tomec@2.2.2.2> doesnt have sips? |
| 17:50.16 | cervajs2 | we have another suspect. pjsip maybe doesnt support savpf |
| 17:52.52 | malcolmd | jameswf: i don't believe that an airport shuttle is included, no. |
| 17:53.04 | malcolmd | so says - http://www.orlandoairports.net/marketing/hotels.htm |
| 17:53.31 | jameswf | https://www.universalorlando.com/Hotels/General-Information/Hotel-Services-Information.aspx <did find this after I asked |
| 17:53.54 | jameswf | Departure Point from Airport: Ground transportation area on lower level of airport. |
| 17:55.30 | malcolmd | oh fun :) |
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| 18:44.20 | cervajs2 | So, pjsip ends SDP negotiation because transport do not match: answer is UDP/TLS/RTP/SAVPF and offer RTP/SAVPF |
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