[Asterisk-Users] SIP newbie question:SIP proxy is necessary?
Jeff
Jeff at mutualphone.com
Mon Feb 9 15:15:42 MST 2004
Hi
a gateway and an ATA186 for test.
1.when I set them up in H323 mode, put gateway's IP in ATA186's
configuration ,ATA186 could send calls directly to gateway, and they
worked just well.
2. after I changed them to sip mode, and ATA186 got no dialtone.
My question is:
In order to send call from ATA186 to gateway,it is necessary to setup a
SIP proxy for ATA186/gateway?
( or it is possible for ATA186 to send call directly to Sip gateway? If
yes, How to setup ATA?)
Jeff
More information about the asterisk-users
mailing list