6 dec 2007 kl. 15.29 skrev Victor Toofic: > El Thu, Dec 06 de 2007 a las 09:53 +0000, Johansson Olle E comentaba: >> What does it mean? Any documentation anywhere on how to implement it? >> Is the previous duration of 2600 for signal 1 cut off to 305? > > The only thing I've found is in this document: > > http://www.africamovies.co.uk/AgentForm/Agent.pdf > > in "3.3.2.3.1.3 Alternatives to RFC 2833" in the section about Sonus > "(c) Sonus". > > Im not sure how that could be implemented. In the case of SIP-to-SIP > my > first thought was that it could be simply relayed, but since I dont > know > the Asterisk's internals I dont know if that is possible or correct.
Asterisk is a multiprotocol PBX, not a SIP proxy. We never just relay something :-) /O _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
