Sangoma Connect works perfectly - until i close the app

recently set up sangoma connect for my extension so i can mobilize better. setup was mostly trouble free (outside of having to reboot the system before i could get service to start) and so was getting it deployed to my phone, which is running android 12. while the app is open everything is honky dory and calls are crystal clear both on wi-fi and cellular. however, if i go to the home screen or any other app, that’s where the issue is. the notification in the notification bar will progress through the following:
-signing off
-disappears altogether and incoming calls don’t ring.
if i open the app again it’ll say “push handshake” for a little bit with the green light blinking, and then everything will function again.

i installed the app and allowed all the permissions they asked of me. i also checked and everything is enabled: display over other apps, notifications, phone, microphone, contacts, as well as battery optimization set to unrestricted. how do i make it run in the background though!? any thoughts would be apprecated.


I would check and confirm that the push servers are getting registered with your PBX when the app is closed/in the background. You may check the registered contacts in Asterisk Info when you have closed the app. There should be a registration coming from an external IP with your extension #.

do i need to open any ports on the gateway? it was my understanding that i don’t. just want to confirm.

SIP signaling port(s) must be open to both the client source IP as well as the full list of cloud servers. Technical Details - Sangoma Connect - Documentation

hey lorne, thanks for dropping by. let me get this right before i march against the gateway. you said

by this you mean the mobile phone must be able to connect to the PBX from both wi-fi and cellular, correct?

this list of cloud servers. can i get that from the automatically added list in the firewall>networks page?

lastly, the SIP signaling ports is merely (in this case) the PJSIP which is default at 5060, and not the media ports, correct?

Obviously for any client to register, it must have access to the SIP port for registration. If you register from both WIFI and GSM then you need open routes for both.

From there or from the wiki page linked above.

Media ports have nothing to do with signaling. I would expect you must have the full RTP range forwarded already, but if not you can expect to have 1-way/0-way audio in some cases.

works perfectly. thanks for the help. looking forward to testing it over time.

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