d67dfea213
SIP is pretty much terrible at dealing with NAT. Asterisk has some knobs that can be enabled to help deal with common issues. Turn them on. In passing, remove the videosupport=yes line. We don't have video enabled for the conference application and we don't have any video codecs allowed, so this option didn't do anything. Change-Id: Ibc17ad3da9bbc110a8cb19daaea1655d0a208670
33 lines
698 B
Plaintext
33 lines
698 B
Plaintext
; sip.conf customizations for pbx.openstack.org
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[general](+)
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udpbindaddr=::
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tcpenable=yes
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tcpbindaddr=::
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context=public
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allowguest=yes
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disallow=all
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allow=g722
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allow=ulaw
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allow=alaw
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allow=gsm
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nat=force_rport,comedia
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<% sip_providers.each do |provider| -%>
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register => <%= provider['username'] %>:<%= provider['password'] %>@<%= provider['hostname'] %>:5060
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<% end -%>
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<% sip_providers.each do |provider| -%>
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<% if provider['outgoing'] == true -%>
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[<%= provider['username'] %>]
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context=public
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host=<%= provider['hostname'] %>
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secret=<%= provider['password'] %>
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type=peer
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defaultuser=<%= provider['username'] %>
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fromuser=<%= provider['username'] %>
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trustrpid=yes
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sendrpid=yes
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<% end -%>
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<% end -%>
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