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LECSJH

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Posts posted by LECSJH

  1. 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?

  2. 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ⓘ

     

  3. 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

  4. 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?

  5. 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?

  6. 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.

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