If you're configuration doesn't allow for these ports, that would be a possible cause of this problem." Other calls were connecting on RTP ports such as: 54988, 19244, 25050 " OK, so your call to 833XXXXXXX established RTP on port 9026.
i also brought to their attention that i was able to place a call to another skyetel number with no issue. I had them go over Wireshark capture and they said they do not see anything out of the ordinary. i have 1 other system, where Skyetel is the provider and calls work to that number both ways with no issues. The provider is Skyetel and they insist the issue is not with them. You have successfully set up your own zones If you want to make one of these zones the default for other interfaces, remember to configure that behavior with the -set-default-zone parameter: sudo firewall-cmd -set-default-zonepublicweb.
#Setup firewall for onsip full#
no changes were made to my knowledge, or that i can find.Īll the proper ports are being forwarded to the PBX and firewall check is 100% ok full cone. sudo firewall-cmd -zone privateDNS -list-services. This is a fairly new system and for the first couple of weeks everything worked fine. inbound calls work absolutely fine across all phones. Outbound calls simply have NO audio from either side. and tried everything within my knowledge and cannot figure this out. ive gone over multiple threads here at 3cx and elsewhere. Sometimes, there's nothing better than a good backtrace on a core dump file to figure out just where the error was raised.Īnyways, I'm looking forward to hearing if those last steps that Tamme sent are working out.Ive been fighting this for going on 3 days. There is a browser client for FreeNode, and the channel has a queue for questions (~take-a-number ). The core developers hang out there most of the day and answer questions pretty promptly. We also tend to be on the FreeSWITCH IRC a lot (#freeswitch on freenode). Tamme linked to this earlier, but the FreeSWITCH wiki has lots of good information on configurations and dialplans. But I wanted to throw a few FreeSWITCH-specific resources in, in case we go beyond our know-how and for future reference to any SIP.js+FreeSWITCH users coming across this issue. We use FreeSWITCH internally at OnSIP, so we know a lot of tricks in debugging issues with it. I'm guessing you are still running through the steps that Tamme listed above. Okay, so since I caused a bunch of emails to be sent anyways, I may as well comment what I was going to say. If you continue to have issues, please get an actual core file so that we can get a legitimate backtrace on what is going on.
#Setup firewall for onsip install#
I DID also install the optional deps for freeswitch:Īgain - this was on a CentOS digital ocean VM with 512mb ram. I basically followed the guide on:īut I replaced the version with v1.5.12. We were then able to connect to the server with the sipjs demo phone,, and make a call to the default dialplans 9196 extension which is a reltime ECHO test. I then edited sofia_profiles/internal.xmlĪnd un commented param name="ws-binding" value=":5066" and changed the port to port 80. Make cd-sounds-install & make cd-moh-install I edited nf and commented out mod_enum because I do not have the ldns-dev lib installed. It measures the jitter and packet loss of your connection, both of which can have a significant impact on VoIP calls. This test will tell you how long it takes for internet traffic to travel between your computer and another device on the internet.
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