1. ybenelli

    [SIP Call] inbound voice 3-4sec delay

    Hi, we have AWS WCS 5 small and we having issues with sip calls, there is a 3 or 4 seconds delay on inbound voice, there is not problem with outbound voice The quality is good, the only problem is the incoming voice delay. The WCS and PBX are on same region, the problem is the same on...
  2. E

    Кодек WebRTC

    Добрый день. У нас появились проблемы со звуком с Манго пришел такой ответ. подскажите пожалуйста, в каких настройках можно отключить этот кодек
  3. H

    Phone SIP Video IOS/Safari macos can not get speaker (audio output)

    Hi admin, When I use safari on Mac/Iphone. I can not get speaker list (audiooutput) So how i can play audio voice in that environment ? Thanks.
  4. H

    Issue Video Call Camera (SIP)

    When i make a call video SIP. Source make call have camera (Phone, Laptop, ....) Destination not have camera (PC) My question: How i get a call just 1 video from source and no video in destination (just audio) Is that possible? Tks.
  5. H

    [Phone Video SIP] How to send and catch custom params

    Hi, I used createCall options "custom" and "sipHeaders" to make a call to another extension PBX (911). But in websocket of 911 i can not found that params. Any thing wrong? Please help me how to send custom params. Thanks.
  6. M

    SIP video calls recording between 2 SIP accounts

    Hi administrator, I want to know WCS can record SIP video calls between 2 SIP accounts. I see a topic in 2017 with the title "SIP calls recording between phone video application" with an answer don't support it but now 2023 WCS can support it? Thank you for reading.
  7. H

    Question about recording video when use Phone Video WebSDK SIP

    Hi, when i use Phone Video SIP I need to record (audio and video or 1 of them). Can WebSDK support function to custom this. And how to setting record this in WCS ... Thanks!
  8. M

    IOS SDK doesn't disconnect calls terminated by the called party

    Hi all, we are using SDK IOS 2.6.105. Our app is a dialer that calls to normal phones dialing the MSISDN (que use asterisk connected with a trunk to an SBC to deliver this kind of calls). We don't have problems with the audio stream in both direcctions, however, when the called party hangs up...
  9. M

    multiple calls from the same number

    it is possible with flashphoner to make multiple calls, to several phone numbers, from the same account, that is the same number, simultaneously. I tried with Zoiper and it works but not with flashphoner.
  10. M

    Can connect to kamailio wss from Phone video of Flashphoner?

    About feature phone video of Flashphoner, I can change WCS URL to my wss server? I had changed but failed, I can't see log failed in server but error in dev tool in browser is "WebSocket connection to 'wss://my_wss_server' failed: Error during WebSocket handshake: Unexpected responese code: 400 "
  11. kevins

    Join Webex Meeting and Republish as RTMP

    Hello, We cannot figure out how to join a Webex, or zoom, meeting from WCS5. What value needs to be provided for the Callee? Also what setting are required for the sip server and proxy? Are these required as WCS5 doesn't register with a sip server in this case, just join as a sip endpoint...
  12. M

    Demo License

    I'm using a demo license to test Web Call Server 5 on Centos Linux Server. I'm wondering if this license allow only one call at time, because i'cant call two numebers with two phone number from messagnet at the same time.
  13. P

    sip problems

    Hello, We have some sip problems couldn't find any solution since 2 months. The problems are: We are using flashphone in our crm. So it is using for mostly outcalls and in an hour an user can make more than 50 calls. 1- without any reason after 2-3 calls while making a new call outcall status...
  14. P

    Asterisk with flashphoner simultaneous extention call not working and outbound call ring tone not playing

    Hello, We are using flashphoner with asterisk freepbx. We have two majer problem: 1. When extentions try to call eachother over flashphoner 2 or more simultaneous connection can't establish. Just one call can establish and others get not extention avaliable response. We tried similar scenario...
  15. M

    audio issue on FlashphonerWebCallServer-5.2.1146

    hello, i have recently installed FlashphonerWebCallServer-5.2.1146 and when i make SIP call using phone-UI . and i make or recevie the call the other party does not hear me. please can you help me to fix this issue. thanks
  16. M

    Prometheus SIP metrics details

    Hi, we are integrating our WCS with Prometheus. I've been searching for the detail of the metrics , I mean, if the values are gauges, counters, etc, but I didn't found toomuch info. Most of all we are interested in SIP calls monitoring , and I see these values: -----Call Stats-----...
  17. M

    SIP phone is working, Video Demos are not

    Hello! Since ages I am using the WCS for managing SIP phone calls. Also on the default demo dashboard I can use the SIP phone demo successfully. Also for ages I did not use any video streaming with the WCS (I think I did it successfully some years ago but I am not totally sure). But for a new...
  18. N

    WCS SIP звонки перестали работать

    Имеется свой SIP-сервер, он рабочий 100% (проверено на Zoiper 2.8.15). После переноса WCS на другой сервер - он перестал звонить. Номера коннектятся к серверу, как видно на прикрепленном скриншоте (и у номера появляется sip contact), но когда пытаешься позвонить - не работают звонки. WCS...
  19. J

    Inbound Call Issues

    I'm trying to use the SIP Flashphoner as a 3CX Client and it can't answer the external Inbound call from Telco. Look like the cause comes from the INVITE send to Flashphoner that doesn't have SDP. But It worked pretty well when I tried to establish a call from the Internal extension with the...
  20. da6hkin

    Ошибка при принятии звонка

    Совершать входящие звонки с веб-клиента на MicroSIP получается, а принимать исходящие не получается. Вылетает ошибка "No remote sdp available". Используем Asterisk.