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PortSIP PBX for Unified Communications

Kristof Sztaho

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  1. We're managed to find to source of the problem. In one of our apps on android the voice recording permission was not granted. Everything works fine now.
  2. Dear PortSIP Team, Thank you for your answer. We've installed PortSIP PBX for this investigation purpose. We've experienced the same issue. The call had a configured sound channel the voice was hearable only at one end. Additionally, the recording wav file contains no understandable voice. Unfortunately we don't have any server-side log from PBX. However we set loglevel to debug, the whole log section is empty. Currently we're investigating. Could you please help me with enabling server-side logging in PBX? I attached the voice recording file of our last call. (It's a .wav file, only I renamed it to mp4 to pass the upload filter). 316061201_7SXD19oMZYXs9nDacvI4vQ_05122020_141424.wav.mp4
  3. Dear PortSIP team! I'd like to ask your help with the following problem: In our currently developed xamarin mobile application we can't hear any voice neither in at the caller nor the called endpoint. We tried it on android platform (with version 9 and 10). After checking the freeswitch logs and the network traffic we found that the call session is initialized correctly and the voice is transferred too. Only we have to amplify the voice volume to a multiplied level (e.g. 100x) to hear anything. In the caller clients we haven't set anything from default regarding to audio settings. The voice of the freeswitch server menu (i.e. dialing tone, message recorder) is audible well. Please help me with a solution in this matter! Thank you!
  4. Hello, We'd like to use a Freeswitch server as SIP communication server connecting to our mobile client apps directly. Regarding to your web site Freeswitch is a supported server component to your VOIP SDK. In our test environment we faced the following problem. After the successful connection between the two mobile clients and the call initialized the call is disconnected after a certain timeout. After examining the Freeswitch output and the Freeswitch documentation the cause was revealed: Aduio connection initialization failure. Freeswitch needs a large open port range (16384-32768 UDP). With the server located in a protected NAT environment forwarding this range to one server is not a suitable solution. Could you provide us a safe and working configuration example of that setup? Currently the only goal is to let two clients from the public internet to connect and speak with each other. (We also see that Freeswitch has NAT traversal options using a STUN server, however configuring a STUN server is not straightforward to us at the moment. If this is the only way: Could you help us with information on STUN configuration?) Thanks in advance! Regards, Kristof
  5. Hi, in our case the onInviteFailure was not called, only the two events described above. It's a bit strange anyway why the onInviteConnected was called in the first place when an explicit reject was answered?
  6. Hello, We're just about to use PortSIP SDK in our new mobile applications (Android and iOS). We faced the following problem in our POC phase: We cannot identify the positive/negative answer from the called client in the caller app. In our caller application (and in the downloadable portsip client app as well) the same onInviteAnswered and onInviteConnected events are triggered on answering and rejecting the call. The only difference in the two scenarios is a tiny difference in the sipMessage: On "answered" case the \"Outbound Call\" string value appears in the sip message in a certain place. When the called number was "offline": application/sdp\r\n appears. The reject and the hangup cannot be identified from each other. Could you please help me with some details of the correct call-handling? Thanks in advance! Regards, Kristof
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