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How to disable icesupport in Asterisk?
This is for customer using Asterisk flavored PBX. icesupport is enabled by default that causes the call to hangup right after 200 OK. As of writing of this document, DIDForSale registered sip trunk with asterisk does not support icesupport.
Ice support can be disabled in /etc/sip.conf.
ssh login in asterisk.
Save and quit.
asterisk -rx "reload"
Some PBX’s might have this setting in sip_customer_general.conf file
Are you having an audio issues in your Asterisk?
Well it’s a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks
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