SIP Authentication methods available?
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Posted by Ramon F Herrera on February 7, 2007, 4:51 pm
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I have an AS-5300 used by several SIP phones (mostly through Asterisk) to dial from VoIP-land to POTS-land. My current "authentication" method is rather draconian, poor and inflexible: I just played with the routing tables to make sure that the only phones that can have IP connectivity with the voice gateway are those in certain subnets. Now, I would like to move up to something that provides better granularity. I understand that Radius is an option? There is also an md5 digest authentication method which may or may not be supported by the Cisco gateway? I am guessing that another (very inefficient) solution would be to set the 'canreinvite=no' option in Asterisk, thereby forcing it to be in the middle of all calls. That would work, right? This solution really sucks. Can somebody knowledgeable discuss and/or point to some documentation on this topic? TIA, -Ramon F Herrera ps: my phones are Linksys/Sipura. Tips on configuring them for authenticated calls are most welcome. | ||||||||||
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Posted by Hongtian on February 8, 2007, 12:02 am
Please log in for more thread options Most Cisco gateway can support MD5 digest authentication. If you have a MS-windows platform, maybe you can try miniSipServer. It is also support this authentication and very easy to use. On 2=D4=C28=C8=D5, =C9=CF=CE=E75=CA=B151=B7=D6, "Ramon F Herrera" <r...@con= exus.net> wrote:  | ||||||||||
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> to dial from VoIP-land to POTS-land. My current "authentication"
> method is rather draconian, poor and inflexible: I just played with
> the routing tables to make sure that the only phones that can have IP
> connectivity with the voice gateway are those in certain subnets.
> Now, I would like to move up to something that provides better
> granularity. I understand that Radius is an option? There is also an
> md5 digest authentication method which may or may not be supported by
> the Cisco gateway?
> I am guessing that another (very inefficient) solution would be to set
> the 'canreinvite=3Dno' option in Asterisk, thereby forcing it to be in
> the middle of all calls. That would work, right? This solution really
> sucks.
> Can somebody knowledgeable discuss and/or point to some documentation
> on this topic?
> TIA,
> -Ramon F Herrera
> ps: my phones are Linksys/Sipura. Tips on configuring them for
> authenticated calls are most welcome.