LECSJH

When I attempt to play call recordings inside the apps, it doesn't play the recording. Checking inside debugger, it gets a 404 error. GET https://pbx.domain/rest/user/102@tenant.domain.com/recs?id=1000 404 (Not Found) (although in the admin panel, https://pbx.domain.com/rest/domain/tenant.domain.com/recs?id=1000 does work.) I am also seeing this inside of the debugger, but maybe less important. DevTools failed to load source map: Could not load content for https://tenant.domain.com/libraries/pdf.js.map: HTTP error: status code 404, net::ERR_HTTP_RESPONSE_CODE_FAILURE The webserver on debug 9 only shows the following: Request from xx.xx.xx.xx:37849 for /rest/user/102@tenant.domain.com//recs?id=1000ⓘ GET /rest/user/102@tenant.domain.com/recs?id=1000 HTTP/1.1 Host: tenant.domain.com Connection: keep-alive sec-ch-ua: "Chromium";v="106", "Google Chrome";v="106", "Not;A=Brand";v="99" Accept-Encoding: identity;q=1, *;q=0 sec-ch-ua-mobile: ?0 User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/106.0.0.0 Safari/537.36 sec-ch-ua-platform: "Windows" Accept: */* Sec-Fetch-Site: same-origin Sec-Fetch-Mode: no-cors Sec-Fetch-Dest: audio Referer: https://tenant.domain.com/usr_portal.htm Accept-Language: en-US,en;q=0.9 Cookie: session=68i1gtd3gqovslie1s2m Range: bytes=0- [9] 0:00:33.984 Last message repeated 2 timesⓘ [8] 0:00:33.984 REST: GET /rest/user/102@tenant.domain.com/recs?id=1000ⓘ [8] 0:00:33.984 Last message repeated 2 timesⓘ [8] 0:00:33.984 REST: Return 404 Not Foundⓘ Any clue where I could start to debug this?

If I wanted to download the beta version, how would I go about doing that?

Oh neat! I'll keep an eye out for it.

ETA of release? The use of OPUS would be fantastic, especially for mobile app users.

A log level 9 with only Media events of during the call with a DTMF 2 being pressed. [4] 22:44:20.230 Last message repeated 4 timesⓘ [6] 22:44:20.230 Port 271: Allocating port for SIP Call-ID e4624c61@pbxⓘ [7] 22:44:20.230 Port 271: SRTP tx keys: pTMcOzZqTnGl4CARsWO8sHfxjWOfwiOYhqypZjvP AA02E18Eⓘ [7] 22:44:20.231 Port 271: Allocated ports 59466 and 59467ⓘ [8] 22:44:20.231 Port 271: Added predefined codec 6 (mapped to 9)ⓘ [8] 22:44:20.231 Port 271: Added predefined codec 2 (mapped to 0)ⓘ [8] 22:44:20.231 Port 271: Added predefined codec 3 (mapped to 8)ⓘ [8] 22:44:20.231 Port 271: Added predefined codec 9 (mapped to 13)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 8 (mapped to 111)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 15 (mapped to 103)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 16 (mapped to 104)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 17 (mapped to 106)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 18 (mapped to 105)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 19 (mapped to 110)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 20 (mapped to 112)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 21 (mapped to 113)ⓘ [8] 22:44:20.231 Port 271: Added rtpmap codec 1 (mapped to 126)ⓘ [7] 22:44:20.234 Port 271: Set codec preference count 5ⓘ [6] 22:44:20.234 Port 272: Allocating port for SIP Call-ID 9abc7c31@pbxⓘ [7] 22:44:20.234 Port 272: SRTP tx keys: 1lnoEih28yogD2I1eQzuxNW5vwJAa6W4pazQodaA 0BD9AC23ⓘ [7] 22:44:20.234 Port 272: Set codec preference count 5ⓘ [8] 22:44:20.234 Port 272: state code from 0 to 100ⓘ [9] 22:44:20.234 Port 272: Adding codec opus/48000 to available listⓘ [9] 22:44:20.234 Port 272: Adding codec PCMU/8000 to available listⓘ [9] 22:44:20.234 Port 272: Adding codec G722/8000 to available listⓘ [9] 22:44:20.234 Port 272: Adding codec G729/8000 to available listⓘ [9] 22:44:20.234 Port 272: Update codecs preference size 5, available codecs size 5ⓘ [7] 22:44:20.235 Port 272: Allocated ports 57278 and 57279ⓘ [8] 22:44:20.236 Port 271: state code from 0 to 183ⓘ [8] 22:44:20.236 Port 271: Ignore double SDPⓘ [9] 22:44:20.237 Port 271: Adding codec opus/48000 to available listⓘ [9] 22:44:20.237 Port 271: Adding codec PCMU/8000 to available listⓘ [9] 22:44:20.237 Port 271: Adding codec G722/8000 to available listⓘ [9] 22:44:20.237 Port 271: Connected device does not support codec G729/8000ⓘ [9] 22:44:20.237 Port 271: Update codecs preference size 5, available codecs size 4ⓘ [6] 22:44:20.237 Port 271: Choose codec opus/48000ⓘ [6] 22:44:20.395 Port 271: Sending RTP to 312.321.321.321:24798, codec opus/48000ⓘ [7] 22:44:20.572 Port 271: Set DTLS SRTP key for clientⓘ [7] 22:44:20.573 Port 271: SRTP tx keys: onsAyI4tkuBFvR1I1K+a4EP2gP3CmKr2LTyQe4BS 1C38146Aⓘ [7] 22:44:20.573 Port 271: SRTP rx keys: Fr7d0ZAt2l5RZLozRGA9FTlKon6iJgnbLtjunBG0 00000000ⓘ [9] 22:44:20.594 Port 271: Received first RTP packetⓘ [8] 22:44:21.545 Port 272: Added predefined codec 2 (mapped to 0)ⓘ [8] 22:44:21.545 Port 272: Added rtpmap codec 1 (mapped to 101)ⓘ [7] 22:44:21.545 Port 272: Set packet length to 20ⓘ [6] 22:44:21.546 Port 272: Choose codec PCMU/8000 in answerⓘ [6] 22:44:21.546 Port 272: Sending RTP to 10.0.1.167:21834, codec PCMU/8000ⓘ [7] 22:44:21.546 Port 272: Determine pass-through mode after receiving responseⓘ [8] 22:44:21.547 Port 272: state code from 100 to 200ⓘ [8] 22:44:21.547 Port 271: state code from 183 to 200ⓘ [7] 22:44:21.548 Port 271: RTP pass-through modeⓘ [7] 22:44:21.548 Port 272: RTP pass-through modeⓘ [7] 22:44:21.548 Port 272: Media-aware pass-through modeⓘ [8] 22:44:21.554 Media: Dropping audio_uk/mb_no_name_ask1.wav from cacheⓘ [6] 22:44:21.584 Port 272: Sending RTP to 123.123.123.123:21834, codec PCMU/8000ⓘ [9] 22:44:21.584 Port 272: Received first RTP packetⓘ [6] 22:44:21.584 Port 271: Different Codecs (local PCMU/8000, remote opus/48000), falling back to transcodingⓘ [9] 22:44:22.732 Port 271: RTCP SR time=3875222662:3819711984 timestamp=2850696874 packets=107 octets=16820ⓘ [9] 22:44:25.562 Port 272: RTCP SR time=3875222665:2390673286 timestamp=32800 packets=199 octets=31840ⓘ [9] 22:44:27.297 Port 271: RTCP SR time=3875222667:1958822914 timestamp=2850916090 packets=336 octets=51162ⓘ [9] 22:44:29.582 Port 272: RTCP SR time=3875222669:2476478142 timestamp=64960 packets=400 octets=64000ⓘ [9] 22:44:33.602 Port 272: RTCP SR time=3875222673:2562343129 timestamp=97120 packets=601 octets=96160ⓘ [9] 22:44:33.873 Port 271: RTCP SR time=3875222674:133234180 timestamp=2851231690 packets=664 octets=103760ⓘ [9] 22:44:37.393 Port 271: RTCP SR time=3875222677:2371818379 timestamp=2851400698 packets=840 octets=132096ⓘ [9] 22:44:37.622 Port 272: RTCP SR time=3875222677:2648233885 timestamp=129280 packets=802 octets=128320ⓘ [9] 22:44:41.551 Port 271: RTCP SR time=3875222681:3053700272 timestamp=2851600378 packets=1048 octets=165584ⓘ [7] 22:44:42.392 Port 271: Received RFC4733 DTMF on codec 126ⓘ [9] 22:44:42.642 Port 272: RTCP SR time=3875222682:2734163295 timestamp=169440 packets=1053 octets=168480ⓘ [9] 22:44:47.051 Port 271: RTCP SR time=3875222687:906332588 timestamp=2851864330 packets=1324 octets=209545ⓘ [9] 22:44:47.662 Port 272: RTCP SR time=3875222687:2820028281 timestamp=209600 packets=1304 octets=208640ⓘ [8] 22:44:50.812 Port 272: Clearing port with SIP Call-ID 9abc7c31@pbxⓘ [8] 22:44:50.833 Media: File recordings/xxxxxxxxxxx/102/20221019-224421-o-102.wav has been writtenⓘ [8] 22:44:50.902 Port 271: state code from 200 to 486ⓘ [8] 22:44:50.902 Port 271: Send hangup with reason byeⓘ [8] 22:44:50.971 Port 271: Clearing port with SIP Call-ID e4624c61@pbxⓘ

We are seeing it inside our Vodia web client. Here is a capture using G.711U (IPs anonymized) https://vm.lecsvoip.com/cdr.php?cdrId=5369381&anonIps=1&hash=c4117d7c26025aedb291edf108285f95ce3dd9716503348d0839f57184b97276 Here is a our capture using OPUS (IPs anonymized)(IPs anonymized) https://vm.lecsvoip.com/cdr.php?cdrId=5369419&anonIps=1&hash=d4ba553883edfc35afe6c866267ac0f29ad5372b2cf32a14422567c315b8cbd7

I can't seem to get our Yealink RPS integration working. I have entered our API credentials that we generated at: https://dm.yealink.com/manager/systemManage/apiService Maybe this is the wrong place to generate API credentials for the RPS? When I enter those into Vodie, and attempt to sync all mac addresses, I see this is our log files: {"data":null,"error":{"errorCode":401,"fieldErrors":[],"msg":"platform.not.available"},"ret":-1} and Server: nginx Date: Tue, 18 Oct 2022 20:05:39 GMT Content-Type: application/json;charset=UTF-8 Transfer-Encoding: chunked Connection: keep-alive Strict-Transport-Security: max-age=16000000;includeSubDomains;preload; X-Frame-Options: DENY Referrer-Policy: no-referrer-when-downgrade X-Content-Type-Options: nosniff X-XSS-Protection: 1;mode=block Maybe our Distributor hasn't given us full permissions for API?

Am I dumb and missing how to start a new SMS conversation inside of the iphone app? Obviously I can add them as a contact, and then start a conversation. But what if my user wants to send a quick one of message without adding the person as a contact?

We are increasingly getting more and more request for inbound SMS to group. An inbound SMS being routed to multiple people at the same time would be beneficial. I understand tracking SMS with previous conversations, but I feel like there are several solutions that can easily track it. Also, allowing agents to respond to that message sent to the queue, as the queue's ANI would be beneficial as well.