![]() I have a suspicion I should choose a fixed RTP port (Preferences > Advanced) that is in the 20001-30000 range but, again, personal testing seems to confirm an out of the box setup works. It registers and dials fine so it seems like SIP signalling is fine, it's just an issue with RTP audio. This is set up to connect to our ip address and nonstandard SIP port 5070 and it's set to use the default STUN service,. ![]() For our test, lets consider an out-of-the-box Zoiper install. I've had no luck swapping out softphones. Two thoughts come to mind server side: Should I be setting my trunk services to type=friend? And are my NAT related configuration settings correct and in the right place? Especially canreinvite=nonat and nat=force_rport,comedia ![]() This avoids connection spamming from foreign robo dialers looking for a free ride.īelow is an excerpt from Asterisk's sip.conf (specific details redacted, you may need to scroll this sample) I am baffled.Īlso be aware that I've tried on a mix of employee systems, both Mac and PC, and I've open ports on their firewalls, turned firewalls completely off, and so on.įor testing purposes, I have no rules in IPTables and the server is not behind our local router's NAT (consumer routers call this the DMZ).Īsterisk configuration is vanilla out of the box with two exceptions: SIP bind port is 5070 and RTP port range is 20001-30000. I've also tested a softphone from my home connection, I've used VPN to tunnel through and connect through countries half way across the globe to check for latency issues, and I've had a family member who lived miles away test and they were successful. 100% of employee's softphones work locally in the office. The classic one-way audio issue that plagues SIP. The person on the land line will not receive any audio from the sip softphone. When a SIP softphone from a work-at-home employee attempts to make a call to, say, a land line, the sip softphone will receive audio but not send it. Suffice to say I have little knowledge of IP telephony. The server is set up for SIP phones to use our SIP trunk via voip.ms/didlogic (two different services because we like international rates on didlogic, but it doesn't have all the domestic features voip.ms has Not relevant to the issue at hand though). Running Ubuntu 14.04 LTS and Asterisk 11.7.0.
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